August 2004 Archives by date

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Starting: Sun Aug 1 00:55:36 MST 2004
Ending: Tue Aug 31 23:18:36 MST 2004
Messages: 3747

• [Asterisk-Users] X100P wants to use g2   Chad Brown
• [Asterisk-Users] learning from the audio folks   Florin Andrei
• [Asterisk-Users] Hiring Setup   Sunrise Ltd
• [Asterisk-Users] MGCP & Cisco ATA 186 Help   Florian Overkamp
• [Asterisk-Users] Zaptel - incoming delay   George Gardiner
• [Asterisk-Users] Cisco 7960 backlight update and prices.   asteriskstuff at ziplip.com
• [Asterisk-Users] 480i User Feedback With Asterisk 802.1Q?   jparr at bgcfreedom.com
• [Asterisk-Users] Cisco 7960 backlight update and prices.   Kevin Walsh
• [Asterisk-Users] X100P wants to use g2   Kevin Walsh
• [Asterisk-Users] Zaptel - incoming delay   Robie Basak
• [Asterisk-Users] Zaptel - incoming delay   Kevin Walsh
• [Asterisk-Users] VoiceMail Not releasing   Kevin Walsh
• [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?   Rich Adamson
• [Asterisk-Users] Does anyone know how to use the DND feature oc Cisco 7940/7960   Daniel Niasoff
• [Asterisk-Users] One More IP Phone for interoperability with Asterisk   Philipp von Klitzing
• [Asterisk-Users] Using a DNS name for externip in sip.conf   Ian Hailey
• [Asterisk-Users] distinctive ring on SNOM 200   Dr. Michael J. Chudobiak
• [Asterisk-Users] cdr record for recording location   Nick Cobley
• [Asterisk-Users] Asterisk on Sparc64   Steve Kennedy
• [Asterisk-Users] Does anyone know how to use the DND feature oc Cisco 7940/7960   Kevin Walsh
• [Asterisk-Users] asterisk-oh323-0.6.3a   Michael Manousos
• [Asterisk-Users] Snom 220   Kanwar Ranbir Sandhu
• [Asterisk-Users] Asterisk on Sparc64   niles at atheos.net
• [Asterisk-Users] Asterisk scalability?   Roy Sigurd Karlsbakk
• [Asterisk-Users] distinctive ring on SNOM 200   Steve Szmidt
• [Asterisk-Users] MGCP & Cisco ATA 186 Help   Dmitri Baranov
• [Asterisk-Users] MGCP & Cisco ATA 186 Help   Dmitri Baranov
• [Asterisk-Users] Best Linux for Asterisk   Ming-Wei Shih
• [Asterisk-Users] Asterisk on Sparc64   Ming-Wei Shih
• [Asterisk-Users] Asterisk on Sparc64   Sunrise Ltd
• [Asterisk-Users] one extention, multiple phones   Chris A. Icide
• [Asterisk-Users] Grandstream Message Waiting light   Steven P. Donegan
• [Asterisk-Users] Asterisk on Sparc64   Sunrise Ltd
• [Asterisk-Users] Grandstream Message Waiting light   Chris
• [Asterisk-Users] Grandstream Message Waiting light   Steven P. Donegan
• [Asterisk-Users] Parking & SIP Phones   Trevor Peirce
• [Asterisk-Users] Writing messages to ISDN phone displays?   Ian Hailey
• [Asterisk-Users] Parking & SIP Phones   jparr at bgcfreedom.com
• [Asterisk-Users] Zaphfc CallerID problem...   Massimo De Nadal
• [Asterisk-Users] Compiling * on OpenBSD 3.5   mpwspam-digiumlist at yahoo.com
• [Asterisk-Users] Digium FXO Interfaces don't support groundstart???   Frank Cofer
• [Asterisk-Users] distinctive ring on SNOM 200   Dr. Michael J. Chudobiak
• [Asterisk-Users] no incoming pstn ring tone   asterisk at bsius.com
• [Asterisk-Users] No ring / Dead Air when transferring from our IVR   asterisk at bsius.com
• [Asterisk-Users] No ring / Dead Air when transferring from our IVR   asterisk at bsius.com
• [Asterisk-Users] Asterisk scalability?   T. Chan
• [Asterisk-Users] Z110p card linh hang up   Martin Keding
• [Asterisk-Users] Asterisk scalability?   James H. Cloos Jr.
• [Asterisk-Users] 480i User Feedback With Asterisk 802.1Q?   Martin Keding
• [Asterisk-Users] distinctive ring on SNOM 200   Steve Szmidt
• [Asterisk-Users] Asterisk scalability?   Scott Stingel
• [Asterisk-Users] distinctive ring on SNOM 200   Dr. Michael J. Chudobiak
• [Asterisk-Users] Message Lamps across IAX connected switches.   jparr at bgcfreedom.com
• [Asterisk-Users] distinctive ring on SNOM 200   Steve Szmidt
• [Asterisk-Users] Collect recording before sending to extension or queue   Greg Boehnlein
• [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *   matt.riddell at sineapps.com
• [Asterisk-Users] Collect recording before sending to extension or queue   Trevor Peirce
• [Asterisk-Users] cdr record for recording location   Leo Ann Boon
• [Asterisk-Users] Best Linux for Asterisk   matt.riddell at sineapps.com
• [Asterisk-Users] SIP- PSTN Gateway   Rudy.Prakash at peregrine-semi.com.au
• [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *   Karl J. Vesterling
• [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *   Billy Huddleston
• [Asterisk-Users] SIP- PSTN Gateway   Andrew Thompson
• [Asterisk-Users] broadvoice/asterisk incoming calls problem   Andrew Thompson
• [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *   Steve Szmidt
• [Asterisk-Users] Re: rate-engine loading error   Glynn Condez
• [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *   Steve Szmidt
• [Asterisk-Users] About CDR billsec when used TE410P   JackyChen
• [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *   Gary
• [Asterisk-Users] Astricon Payment   calvis
• [Asterisk-Users] Sipura 3000 PSTN disconnect in the UK   Dameon D. Welch-Abernathy
• [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?   Dameon D. Welch-Abernathy
• [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *   Sunrise Ltd
• [Asterisk-Users] Zaphfc CallerID problem...   Thorsten Huber
• [Asterisk-Users] Cisco 7960 backlight and list etiquette?   Mitchel Constantin
• [Asterisk-Users] One More IP Phone for interoperability with Asterisk   Holger Schurig
• [Asterisk-Users] One More IP Phone for interoperability with Asterisk   Holger Schurig
• [Asterisk-Users] different pridialplan for different channels in zapata.conf   Key Aavoja
• [Asterisk-Users] Parking & SIP Phones   Jason Williams
• [Asterisk-Users] One More IP Phone for interoperability with Asterisk   Oleg A. Arkhangelsky
• [Asterisk-Users] Parking & SIP Phones   jparr at bgcfreedom.com
• [Asterisk-Users] Parking & SIP Phones   Jason Williams
• [Asterisk-Users] h.323 debug   Alexey Hrapunov
• [Asterisk-Users] detect FAX terminal   bit123
• [Asterisk-Users] sip over h323   Thomas Kuepper
• [Asterisk-Users] Parking & SIP Phones   Steve Szmidt
• [Asterisk-Users] CDR with MySQL and Asterisk PID File   Tom Lawrence
• [Asterisk-Users] CDR with MySQL and Asterisk PID File   Holger Schurig
• [Asterisk-Users] CDR with MySQL and Asterisk PID File   Kevin Walsh
• [Asterisk-Users] CDR with MySQL and Asterisk PID File   Kevin Walsh
• [Asterisk-Users] Zaphfc CallerID problem...   Massimo De Nadal
• [Asterisk-Users] RC1 - error message : Request to schedule in the past   Areski
• [Asterisk-Users] Zaphfc CallerID problem...   Thorsten Huber
• [Asterisk-Users] Win2000 DUN via Asterisk (Is it possible)   Jonathan Moore
• [Asterisk-Users] Fax on demand   Isamar Maia
• [Asterisk-Users] Zaphfc CallerID problem...   Massimo De Nadal
• [Asterisk-Users] Parking & SIP Phones   Nicolas Gudino
• [Asterisk-Users] Win2000 DUN via Asterisk (Is it possible)   Jon Creasey
• [Asterisk-Users] Cisco 7960 backlight update and prices.   asteriskstuff at ziplip.com
• [Asterisk-Users] Zaphfc CallerID problem...   Massimo De Nadal
• [Asterisk-Users] Help with Quicknet PhoneJack@Asterisk   Fritz Reichmann
• [Asterisk-Users] Fax on demand   Brian McManus
• [Asterisk-Users] DID's in the Czech Republic   Steven Kokinos
• [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing   Roman Bessyadovskii
• [Asterisk-Users] Win2000 DUN via Asterisk (Is it possible)   Jon Creasey
• [Asterisk-Users] RTP Packets going over caller and calle !!   Carlos Arnt
• [Asterisk-Users] 480i User Feedback With Asterisk 802.1Q?   Jayson Vantuyl
• [Asterisk-Users] CDR with MySQL and Asterisk PID File   Scott Stingel
• [Asterisk-Users] Clustering in Asterisk   Trilogy India
• [Asterisk-Users] Fwd: Help with Quicknet PhoneJack@Asterisk   Fritz Reichmann
• [Asterisk-Users] Fax on demand   Isamar Maia
• [Asterisk-Users] Astricon Payment   Steven Sokol
• [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn   joachim
• [Asterisk-Users] RC1 - error message : Request to schedule in the past   Greg Hill
• [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing   Ken Godee
• [Asterisk-Users] Cisco MC3810   Wayde Nie
• [Asterisk-Users] Selling asterisk-based solutions   David Gurr
• [Asterisk-Users] (no subject)   Tom Lawrence
• [Asterisk-Users] Multiple Line SIP Phones?   Noah Miller
• [Asterisk-Users] Cisco MC3810   Steve Szmidt
• [Asterisk-Users] Vonage catastrophic failure...   Jay Milk
• [Asterisk-Users] Selling asterisk-based solutions   Dave Weis
• [Asterisk-Users] Stripping characters from SIP dial strings   David Gurr
• [Asterisk-Users] Performance of queues   Florian Overkamp
• [Asterisk-Users] DID's in the Czech Republic   Jan Janak
• [Asterisk-Users] (no subject)   Steven Critchfield
• [Asterisk-Users] RTP Packets going over caller and calle !!   Greg Hill
• [Asterisk-Users] How STUN work?   ShanKutti
• [Asterisk-Users] avm c4, ptmp   Maurizio Marini
• [Asterisk-Users] Performance of queues   Robert Jackson
• [Asterisk-Users] Vonage catastrophic failure...   Sunrise Ltd
• [Asterisk-Users] bri-stuff.0.1.0-RC2k + hfc card: dropouts on IAX2 & MP3Player quits on streams   Deti Fliegl
• [Asterisk-Users] App.c   AJ Grinnell
• [Asterisk-Users] App.c   Steve Hanselman
• [Asterisk-Users] avm c4, ptmp   Deti Fliegl
• [Asterisk-Users] App.c   AJ Grinnell
• [Asterisk-Users] App.c   Seth Remington
• [Asterisk-Users] System Requirements   Brent Franks
• [Asterisk-Users] New CVS and Sipuras   AJ Grinnell
• [Asterisk-Users] How STUN work?   Karl Brose
• [Asterisk-Users] DID's in Spain   Adria Vidal
• [Asterisk-Users] How STUN work?   sgup015 at ec.auckland.ac.nz
• [Asterisk-Users] Asterisk on Sparc64   Ming-Wei Shih
• [Asterisk-Users] Softphone - Freeware?!   Nicolas Gudino
• [Asterisk-Users] System Requirements   Beierlein Moritz
• [Asterisk-Users] New CVS and Sipuras   Kevin Walsh
• [Asterisk-Users] broadvoice/asterisk incoming calls problem   Bartosz Wegrzyn
• [Asterisk-Users] asterisk-oh323-0.6.3a   M. Willigs
• [Asterisk-Users] Today's possible problems with Broadvoice????   Bartosz Wegrzyn
• [Asterisk-Users] Softphone - Freeware?!   Eric Bart
• [Asterisk-Users] Today's possible problems with Broadvoice????   Andrew Kohlsmith
• [Asterisk-Users] Today's possible problems with Broadvoice????   Bartosz Wegrzyn
• [Asterisk-Users] Today's possible problems with Broadvoice????   Bartosz Wegrzyn
• [Asterisk-Users] Softphone - Freeware?!   Nicolas Gudino
• [Asterisk-Users] Today's possible problems with Broadvoice????   Brian McManus
• [Asterisk-Users] Asterisk as Front-End for Artisoft Televantage 6   Alain Bautista
• [Asterisk-Users] Today's possible problems with Broadvoice????   Bartosz Wegrzyn
• [Asterisk-Users] Pre-release of OSX GUI tool to add extensions and phones   Sunrise Ltd
• [Asterisk-Users] How STUN work?   Karl Brose
• [Asterisk-Users] Softphone - Freeware?!   Eric Bart
• [Asterisk-Users] Today's possible problems with Broadvoice????   Brian McManus
• [Asterisk-Users] asterisk call parking + SNOM lighted buttons?   Dr. Michael J. Chudobiak
• [Asterisk-Users] Today's possible problems with Broadvoice????   Andrew Kohlsmith
• [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Chris Shaw
• [Asterisk-Users] First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs?   Brian Hudson
• [Asterisk-Users] Playback doesn't work whith h323   M. Willigs
• [Asterisk-Users] First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs?   Florian Overkamp
• [Asterisk-Users] First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs?   Brian McSpadden
• [Asterisk-Users] First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs?   Brian Elton
• [Asterisk-Users] CallPres screening DDI   ePyron Felix Deierlein
• [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Andrew Kohlsmith
• [Asterisk-Users] CallPres screening DDI   Steven Critchfield
• [Asterisk-Users] Help with ParkAndAnnounce command   Zolt Egeto
• [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Chris Shaw
• [Asterisk-Users] asterisk+radius   mohammad mirzaee
• [Asterisk-Users] CallPres screening DDI   Kevin Walsh
• [Asterisk-Users] One way voice with following error   Owais Zuber
• [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Andrew Kohlsmith
• [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Rich Adamson
• [Asterisk-Users] G729 Codec   Daniel Niasoff
• [Asterisk-Users] G729 Codec   Adam Hart
• [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Chris Shaw
• [Asterisk-Users] MPG123, Music On Hold and Variable Bit Rate   Mike Coakley
• [Asterisk-Users] G729 Codec   Steve Underwood
• [Asterisk-Users] G729 Codec   Steve Underwood
• [Asterisk-Users] G729 Codec   Adam Hart
• [Asterisk-Users] Ringer Equivalencies?   Matt Davies | MattDavies.Net
• [Asterisk-Users] GET VARIABLE with AGI   Arnaud Pignard
• [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Bartosz Wegrzyn
• [Asterisk-Users] Ringer Equivalencies?   Andrew Kohlsmith
• [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Andrew Kohlsmith
• [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Steven Critchfield
• [Asterisk-Users] Ringer Equivalencies?   Matt Davies | MattDavies.Net
• [Asterisk-Users] different pridialplan for different channels in zapata.conf   Steve Totaro
• [Asterisk-Users] Ringer Equivalencies?   Andrew Kohlsmith
• [Asterisk-Users] Fax on demand   Ken Wiesner
• [Asterisk-Users] G729 or GSM   Jean-Yves Avenard
• [Asterisk-Users] Help with Cisco PRI   gavron at wetwork.net
• [Asterisk-Users] Cisco PRI no CallerID   gavron at wetwork.net
• [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Wolfgang S. Rupprecht
• [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Geoff Nordli
• [Asterisk-Users] Ringer Equivalencies?   Kevin P. Fleming
• [Asterisk-Users] help with digium E1 card   Horacio J. Peña
• [Asterisk-Users] help with digium E1 card   Scott Stingel
• [Asterisk-Users] G729 or GSM   Kevin Walsh
• [Asterisk-Users] Making asterisk distributed   Trilogy India
• [Asterisk-Users] G729 Codec+packet loss concealment   clive18 at webmail.co.za
• [Asterisk-Users] Making asterisk distributed   WipeOut
• [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing   Roman Bessyadovskii
• [Asterisk-Users] G729 Codec   steve at daviesfam.org
• [Asterisk-Users] OH323 not dial Modem[i4l]/g1   eltorio
• [Asterisk-Users] Using Clustering/TDMoE   Trilogy India
• [Asterisk-Users] Using Clustering/TDMoE   Steven Critchfield
• [Asterisk-Users] App.c   Jason Williams
• [Asterisk-Users] Asterisk as Front-End for Artisoft Televantage 6   Jason Williams
• [Asterisk-Users] Cisco PRI no CallerID   Jason Williams
• [Asterisk-Users] Cisco PRI no CallerID   Low, Adam
• [Asterisk-Users] avm c4, ptmp   Maurizio Marini
• [Asterisk-Users] G729 Codec+packet loss concealment   Andrew Kohlsmith
• [Asterisk-Users] avm c4, ptmp   Jason Williams
• [Asterisk-Users] Echo problems with mISDN?   CK
• [Asterisk-Users] avm c4, ptmp   Maurizio Marini
• [Asterisk-Users] Called ID in Australia   Robert Barnes
• [Asterisk-Users] chan_sccp2 testers needed   Alexei Chetroi
• [Asterisk-Users] bearer Capability   Thomas Kuepper
• [Asterisk-Users] Called ID in Australia   Kimble Young
• [Asterisk-Users] help with digium E1 card   Horacio J. Peña
• [Asterisk-Users] A few questions   Mark
• [Asterisk-Users] G729 Codec+packet loss concealment   steve at daviesfam.org
• [Asterisk-Users] G729 Codec+packet loss concealment   clive18 at webmail.co.za
• [Asterisk-Users] Called ID in Australia   Christopher Lee
• [Asterisk-Users] Playback doesn't work whith h323   Jeremy McNamara
• [Asterisk-Users] G729 Codec+packet loss concealment   Andrew Kohlsmith
• [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Rich Adamson
• [Asterisk-Users] A few questions   Steven Critchfield
• [Asterisk-Users] Any small colleges/universities using PBX or Voicemail?   Brian Hudson
• [Asterisk-Users] First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs?   Brian Hudson
• [Asterisk-Users] Called ID in Australia   Shaun Ewing
• Fw: [Asterisk-Users] Digium FXO Interfaces don't support groundstart???   Frank Cofer
• [Asterisk-Users] Any small colleges/universities using PBX or Voicemail?   Tony Nichols
• [Asterisk-Users] avm c4: DISCONNECT_IND ID=001 #0x0193 LEN=0014   Maurizio Marini
• [Asterisk-Users] A few questions - isdn call routing   clive18 at webmail.co.za
• [Asterisk-Users] VON Magazine article.   Dave Cotton
• [Asterisk-Users] Called ID in Australia   Adam Goryachev
• [Asterisk-Users] Configure Makefile to run with older iax Protocol   asterisk at gfs-zh.ch
• [Asterisk-Users] features.conf   AJ Grinnell
• [Asterisk-Users] features.conf   Robert Jackson
• [Asterisk-Users] astguiclient: blank php pages   eduardo at garnier.inf.br
• [Asterisk-Users] features.conf   AJ Grinnell
• [Asterisk-Users] astguiclient: blank php pages   mattf
• [Asterisk-Users] DID Trunk   Joe Pukepail
• [Asterisk-Users] Emailing phone messages?   Sean Garland
• [Asterisk-Users] G729 Codec+packet loss concealment   Kevin Walsh
• [Asterisk-Users] PRI Call Redirection / Transfers   John Harragin
• [Asterisk-Users] Hook-flash timing   john lawler
• [Asterisk-Users] Emailing phone messages?   Seth Remington
• [Asterisk-Users] Analog channel stays offhook   Danial Subhani
• [Asterisk-Users] features.conf   Chris Shaw
• [Asterisk-Users] G729 Codec   Steve Underwood
• [Asterisk-Users] ZyXEL 2000w In Call Menu/Hold configs   John Howard
• [Asterisk-Users] G729 Codec+packet loss concealment   Tim McKee
• [Asterisk-Users] VoIP experiences with Cable and DSL   Steve Szmidt
• [Asterisk-Users] G729 Codec   Andres
• [Asterisk-Users] PRI Call Redirection / Transfers   Kris Boutilier
• [Asterisk-Users] G729 Codec   Steve Underwood
• [Asterisk-Users] Rodopi Billing   Darren Bentley
• [Asterisk-Users] features.conf   Kevin Walsh
• [Asterisk-Users] Rodopi Billing   jparr at bgcfreedom.com
• [Asterisk-Users] Best Linux for Asterisk   Leif Madsen
• [Asterisk-Users] features.conf   Josh Roberson
• [Asterisk-Users] G729 Codec   Andres
• [Asterisk-Users] Asterisk Tapi Driver for Windows   Greg Boehnlein
• [Asterisk-Users] Rodopi Billing   Brian D'Arcy
• [Asterisk-Users] PRI Call Redirection / Transfers   Paul Mahler
• [Asterisk-Users] features.conf   Kevin Walsh
• [Asterisk-Users] Asterisk Sighting   calvis
• [Asterisk-Users] features.conf   Josh Roberson
• [Asterisk-Users] Asterisk on Sparc64   Ming-Wei Shih
• [Asterisk-Users] Today's possible problems with Broadvoice????   Jay Milk
• [Asterisk-Users] Integration with Altigen   Geoff Nordli
• [Asterisk-Users] Rodopi Billing   Darren Bentley
• [Asterisk-Users] Asterisk and RT   Justin Carlson
• [Asterisk-Users] Rodopi Billing   Gary Carr
• [Asterisk-Users] Rodopi Billing   Jeremy McNamara
• [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Brian Capouch
• [Asterisk-Users] Re: Integration with Altigen   Jason Kawakami
• [Asterisk-Users] Analog channel stays offhook   steve at daviesfam.org
• [Asterisk-Users] G729 Codec   steve at daviesfam.org
• [Asterisk-Users] Play audio into meetme conference?   Paul Egger
• [Asterisk-Users] Cisco MC3810   Wayde Nie
• [Asterisk-Users] snom 200 - custom melody   Joshua McClintock
• [Asterisk-Users] instable Modem-Module in CVS ?   Christoph Rothe
• [Asterisk-Users] cisco ubr924   Duane Cox
• [Asterisk-Users] Play audio into meetme conference?   Brancaleoni Matteo
• [Asterisk-Users] After RC1 upgrade, temporary loss of voice   asterisk at axelltd.com
• [Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?   Mike Benoit
• [Asterisk-Users] After RC1 upgrade, temporary loss of voice   Steve Szmidt
• [Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?   Andres
• [Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?   Mike Benoit
• [Asterisk-Users] Can Zap detect line is already off-hook?   David Gurr
• [Asterisk-Users] asterisk call parking + SNOM lighted buttons?   Steve Totaro
• [Asterisk-Users] UK VoIP-PSTN gateway recommendations   David Gurr
• [Asterisk-Users] problems with'#' transfer after hold...   Stephen Hon
• [Asterisk-Users] problems with'#' transfer after hold...   Chris Shaw
• [Asterisk-Users] VoIP experiences with Cable and DSL   Andrew Kohlsmith
• [Asterisk-Users] UK VoIP-PSTN gateway recommendations   Scott Stingel
• [Asterisk-Users] VoIP experiences with Cable and DSL   Chris Shaw
• [Asterisk-Users] Rodopi Billing   Ejay Hire
• [Asterisk-Users] VoIP experiences with Cable and DSL   Andrew Kohlsmith
• [Asterisk-Users] Rodopi Billing   Gary Carr
• [Asterisk-Users] Rodopi Billing   Saliel Figueira Filho
• [Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?   Andrew Gordon
• [Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?   Dameon D. Welch-Abernathy
• [Asterisk-Users] Rodopi Billing   Tom
• [Asterisk-Users] VoIP experiences with Cable and DSL   Steve Szmidt
• [Asterisk-Users] New CVS and Sipuras   Trevor Peirce
• [Asterisk-Users] Logging into Multiple Call Queues on two * Servers and Voice Mail option.   Shad Mortazavi
• [Asterisk-Users] Dialplan question   Simon Brown
• [Asterisk-Users] VoIP experiences with Cable and DSL   Chris
• [Asterisk-Users] VoIP experiences with Cable and DSL   Chris
• [Asterisk-Users] VoIP experiences with Cable and DSL   Steve Szmidt
• [Asterisk-Users] VoIP experiences with Cable and DSL   Wolfgang S. Rupprecht
• [Asterisk-Users] AstMan   Wiley E. Siler
• [Asterisk-Users] Gafachi?   Luke Catranis
• [Asterisk-Users] VoIP experiences with Cable and DSL   Chris
• [Asterisk-Users] CID Blocked vs. Unknown   Trevor Peirce
• [Asterisk-Users] VoIP experiences with Cable and DSL   Chris
• [Asterisk-Users] Gafachi?   Luke Catranis
• [Asterisk-Users] CID Blocked vs. Unknown   William Suffill
• [Asterisk-Users] Re: problems with'#' transfer after hold   Sudhir Kumar
• [Asterisk-Users] Hook-flash timing   Adam Goryachev
• [Asterisk-Users] Re: problems with'#' transfer after hold   Chris
• [Asterisk-Users] Very decent book - "VoIP Telephony with Asterisk"   John Vogel
• [Asterisk-Users] PRI Call Redirection / Transfers   Rick L. Wilson, Sr.
• [Asterisk-Users] Cisco MC3810   Gonzalo Gasca Meza
• [Asterisk-Users] CAC AB1 and Asterisk   Ronan
• [Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?   Mike Benoit
• [Asterisk-Users] Rodopi Billing   Greg Boehnlein
• [Asterisk-Users] Call pickup (group)   Florian Overkamp
• [Asterisk-Users] about sip.conf   Murali
• [Asterisk-Users] Cisco SIP Phone 7960 & DTMF Problem   Miroslav Nachev
• [Asterisk-Users] avm c4   Maurizio Marini
• [Asterisk-Users] capturing a call   Maurizio Marini
• [Asterisk-Users] Bearer Capability   Thomas Kuepper
• [Asterisk-Users] rxfax killed asterisk   Vladyslav
• [Asterisk-Users] Asterisk QOS working perfect using sveasoft 3.11g   lists-jmhunter
• [Asterisk-Users] capturing a call   avizion
• [Asterisk-Users] A few questions - isdn call routing   Peter Svensson
• [Asterisk-Users] PRI/H323 gateway   Asmine Ouloube
• [Asterisk-Users] German sounds   Bastian Schern
• [Asterisk-Users] about sip.conf   Duane Cox
• [Asterisk-Users] VoIP experiences with Cable and DSL   Walt Reed
• [Asterisk-Users] German sounds   Christoph Rothe
• [Asterisk-Users] German sounds   Chris
• [Asterisk-Users] Who is calling me ?   Andrei Goncalves
• [Asterisk-Users] Congested link   Alexey Hrapunov
• [Asterisk-Users] 2 sip servers   Altus Snyman
• [Asterisk-Users] No incoming audio on incoming SIP calls   David Gurr
• [Asterisk-Users] VoIP experiences with Cable and DSL   Leif Madsen
• [Asterisk-Users] Who is calling me ?   Leif Madsen
• [Asterisk-Users] A few questions - isdn call routing   clive18 at webmail.co.za
• [Asterisk-Users] SIP pickupgroup   Tais M. Hansen
• [Asterisk-Users] 2 sip servers   Leif Madsen
• [Asterisk-Users] Astetrisk connectet to PBX   Claus Lavdal
• [Asterisk-Users] No incoming audio on incoming SIP calls   Johan Landerholm
• [Asterisk-Users] VoIP experiences with Cable and DSL   Steven P. Donegan
• [Asterisk-Users] Rodopi Billing   Ejay Hire
• [Asterisk-Users] 2 sip servers   Deon Rodden
• [Asterisk-Users] Problems with E100P   Marcelo Rodriguez
• [Asterisk-Users] Logging into Multiple Call Queues on two * Servers and Voice Mail option.   Robert Jackson
• [Asterisk-Users] Re: Analog FXO Card   Francois Menard (Mailing List Account)
• OT: Re: [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Jayson Vantuyl
• [Asterisk-Users] 2 sip servers   Robert Jackson
• [Asterisk-Users] SIP pickupgroup   Holger Schurig
• [Asterisk-Users] 2 sip servers   Leif Madsen
• [Asterisk-Users] 2 sip servers   Altus Snyman
• [Asterisk-Users] Problems with E100P   Jon Stockill
• [Asterisk-Users] PRI Call Redirection / Transfers   John Harragin
• [Asterisk-Users] Making asterisk distributed   Jeremy McNamara
• [Asterisk-Users] IAX2 'no authority found' problem   Simon Ward
• [Asterisk-Users] Rodopi Billing   Jeremy McNamara
• [Asterisk-Users] Barge in on to agents conversation   Navnit Chachan
• [Asterisk-Users] PRI/H323 gateway   Jeremy McNamara
• [Asterisk-Users] rxfax killed asterisk   Andrew Kohlsmith
• [Asterisk-Users] Re: problems with'#' transfer after hold   Andrew Kohlsmith
• [Asterisk-Users] VoIP experiences with Cable and DSL   Andrew Kohlsmith
• [Asterisk-Users] Re: problems with'#' transfer after hold   Chris Shaw
• [Asterisk-Users] Problems with E100P   Scott Stingel
• [Asterisk-Users] Rodopi Billing   Gary Carr
• [Asterisk-Users] Bogus patent applications (was: [RANT] ... Broadvoice)   Sunrise Ltd
• OT: Re: [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????   Steven Critchfield
• [Asterisk-Users] Making asterisk distributed   Steven Critchfield
• [Asterisk-Users] Channel Bank   Joe Pukepail
• [Asterisk-Users] German sounds   Fran Boon
• [Asterisk-Users] Re: IAX2 'no authority found' problem   Tony Mountifield
• [Asterisk-Users] Gafachi?   Charles Ellis
• [Asterisk-Users] IAX2 'no authority found' problem   Josh Roberson
• [Asterisk-Users] Barge in on to agents conversation   Nicolas Gudino
• [Asterisk-Users] Cisco SIP Phone 7960 & DTMF Problem   Nicolas Gudino
• [Asterisk-Users] IAX2 'no authority found' problem   steve at daviesfam.org
• [Asterisk-Users] BT100 bad handset?   Jason Kawakami
• [Asterisk-Users] rxfax killed asterisk   Steve Underwood
• [Asterisk-Users] Get MWI from Telco's voicemail   Scott Petersen
• [Asterisk-Users] Cisco SIP Phone 7960 & DTMF Problem   Miroslav Nachev
• [Asterisk-Users] Rodopi Billing   Darren Bentley
• [Asterisk-Users] Asterisk Integration   felippe at npd.ufsc.br
• [Asterisk-Users] Asterisk & ISDN-card   Evert Meulie
• [Asterisk-Users] problems with'#' transfer after hold...   Wayne
• [Asterisk-Users] Making asterisk distributed   Sunrise Ltd
• [Asterisk-Users] Auto-attendant with an IP trunk   apurohit at verizon.net
• [Asterisk-Users] Cisco SIP Phone 7960 & DTMF Problem   Joshua M. Thompson
• [Asterisk-Users] Cisco SIP Phone 7960 & DTMF Problem   Wayne
• [Asterisk-Users] rxfax killed asterisk   Andrew Kohlsmith
• [Asterisk-Users] Gafachi?   Chris Foster
• [Asterisk-Users] Zultys ZIP2   Asterisk User
• [Asterisk-Users] Re: Integration with Altigen   Geoff Nordli
• [Asterisk-Users] PSTN Access Providers for Asterisk   William R. Lorenz
• [Asterisk-Users] Asterisk Integration   Andrew Kohlsmith
• [Asterisk-Users] calling card on Zap outbound   rich allen
• [Asterisk-Users] Auto-attendant with an IP trunk   Joshua McClintock
• [Asterisk-Users] Rodopi Billing   Gary Carr
• [Asterisk-Users] H323 Call Dropping   Asterisk .
• [Asterisk-Users] RE: No incoming audio on incoming SIP calls   David Gurr
• [Asterisk-Users] Using Cisco SIP Phones with Asterisk   Gary Carr
• [Asterisk-Users] Asterisk & ISDN-card   Martin List-Petersen
• [Asterisk-Users] Problems with E100P   Marcelo Rodriguez
• [Asterisk-Users] Asterisk Integration   Jay Milk
• [Asterisk-Users] PSTN Access Providers for Asterisk   Luke Catranis
• [Asterisk-Users] Re: IAX2 'no authority found' problem   Tony Mountifield
• [Asterisk-Users] CAC AB1 and Asterisk   Ronan
• [Asterisk-Users] VoIP experiences with Cable and DSL   Geoff Nordli
• [Asterisk-Users] Asterisk Integration   Senad Jordanovic
• [Asterisk-Users] Get MWI from Telco's voicemail   Seth Remington
• [Asterisk-Users] Auto-attendant with an IP trunk   Seth Remington
• [Asterisk-Users] Playback doesn't work whith h323   M. Willigs
• [Asterisk-Users] RE: No incoming audio on incoming SIP calls   Luke Catranis
• [Asterisk-Users] problems with'#' transfer after hold...   Chris Shaw
• [Asterisk-Users] Re: Auto-attendant with an IP trunk   apurohit at verizon.net
• [Asterisk-Users] Gafachi?   lists-jmhunter
• [Asterisk-Users] CAC AB1 and Asterisk   Steven Critchfield
• [Asterisk-Users] Integrating an old PBX with Asterisk   Marco Vescovi
• [Asterisk-Users] RE: No incoming audio on incoming SIP calls   Steven Critchfield
• [Asterisk-Users] Snom 200 Programmable Keys   Steve Woolley
• [Asterisk-Users] Get MWI from Telco's voicemail   lists-jmhunter
• [Asterisk-Users] Get MWI from Telco's voicemail   Greg Hill
• [Asterisk-Users] PSTN Access Providers for Asterisk   Greg Hill
• [Asterisk-Users] VoIP experiences with Cable and DSL   Steve Szmidt
• [Asterisk-Users] CAC AB1 and Asterisk   Yifang Dai
• [Asterisk-Users] VoIP experiences with Cable and DSL   Steve Szmidt
• [Asterisk-Users] Using Cisco SIP Phones with Asterisk   Mitchel Constantin
• [Asterisk-Users] Identifying which call an event belongs to   Michael Ulitskiy
• [Asterisk-Users] Get MWI from Telco's voicemail   Scott Petersen
• [Asterisk-Users] BT100 bad handset?   Steve Totaro
• [Asterisk-Users] Get MWI from Telco's voicemail   Mitchel Constantin
• [Asterisk-Users] Rodopi Billing   Greg Boehnlein
• [Asterisk-Users] Snom 200 Programmable Keys   Dr. Michael J. Chudobiak
• [Asterisk-Users] CAC AB1 and Asterisk   Tim McKee
• [Asterisk-Users] Get MWI from Telco's voicemail   Steven P. Donegan
• [Asterisk-Users] Re: SPA-3000 as a regular Asterisk FXO device?   Randy Bush
• [Asterisk-Users] Color in console   Dan Mahoney, System Admin
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   Florin Andrei
• [Asterisk-Users] Identifying which call an event belongs to   Nicolas Gudino
• [Asterisk-Users] CAC AB1 and Asterisk   Ronan
• [Asterisk-Users] Snom 200 Programmable Keys   Dr. Michael J. Chudobiak
• [Asterisk-Users] CAC AB1 and Asterisk   Ronan
• [Asterisk-Users] Can't get T or t option to work with two IAX2 channels.   Rious
• [Asterisk-Users] Re: Re: Integration with Altigen   Jason Kawakami
• [Asterisk-Users] Re: Cisco SIP Phone 7960 & DTMF Problem   Randy Bush
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   Chris Shaw
• [Asterisk-Users] CAC AB1 and Asterisk   Ronan
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   Jay Milk
• [Asterisk-Users] Playback doesn't work whith h323   Jeremy McNamara
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   matt.riddell at sineapps.com
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   C. Johnson
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   Steven P. Donegan
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   Leo Ann Boon
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   Greg Hulands
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   C. Johnson
• [Asterisk-Users] Playback doesn't work whith h323   Seth Remington
• [Asterisk-Users] Configure E1 PRI   Marcelo Rodriguez
• [Asterisk-Users] CAC AB1 and Asterisk   Yifang Dai
• [Asterisk-Users] how to integrate an addon module to asterisk   Sathya Weerasooriya
• [Asterisk-Users] New Head Appears to Break SIP to iConnect   Greg Blakely
• [Asterisk-Users] Color in console   Steven Critchfield
• [Asterisk-Users] how to integrate an addon module to asterisk   Steven Critchfield
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   Wolfgang S. Rupprecht
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   Greg Blakely
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   Steven P. Donegan
• [Asterisk-Users] Using answering machine in my phone   Adam Bezanson
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   Wolfgang S. Rupprecht
• [Asterisk-Users] Re: Asterisk QOS working perfect using sveasoft 3.11g   lists-jmhunter
• [Asterisk-Users] Making asterisk distributed   Stephen Hon
• [Asterisk-Users] Playback doesn't work whith h323   Steve Szmidt
• [Asterisk-Users] Playback doesn't work whith h323   Josh Roberson
• [Asterisk-Users] Asterisk & ISDN-card   Holger Schurig
• [Asterisk-Users] Color in console   Dan Mahoney, System Admin
• [Asterisk-Users] Asterisk QOS working perfect using sveasoft 3.11g   John Baker
• [Asterisk-Users] Color in console   Kevin Walsh
• [Asterisk-Users] RE: No incoming audio on incoming SIP calls   John Howard
• AW: [Asterisk-Users] Integrating an old PBX with Asterisk   Stefan Märkle
• [Asterisk-Users] No incoming audio on incoming SIP calls   Tobias Jönsson
• [Asterisk-Users] RE: No incoming audio on incoming SIP calls   John Howard
• AW: [Asterisk-Users] Integrating an old PBX with Asterisk   Peter Svensson
• [Asterisk-Users] No incoming audio on incoming SIP calls   John Howard
• [Asterisk-Users] QuadBri in NT mode not working.   irmantas.gudelis at vilnius2.balt.net
• [Asterisk-Users] No incoming audio on incoming SIP calls   Tobias Jönsson
• [Asterisk-Users] QuadBri in NT mode not working.   Jens von Bülow
• [Asterisk-Users] QuadBri in NT mode not working.   irmantas.gudelis at vilnius2.balt.net
• [Asterisk-Users] QuadBri in NT mode not working.   Jens von Bülow
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   Steve Totaro
• [Asterisk-Users] FCC Rules VoIP Must Be Tappable   Steve Totaro
• [Asterisk-Users] Using answering machine in my phone   Steve Totaro
• [Asterisk-Users] H323 Call Dropping   Asterisk .
• [Asterisk-Users] Call Transfer Problems with Grandstream Budgetone 100 Phone   James Dutton
• [Asterisk-Users] Call Transfer Problems with Grandstream Budgetone 100 Phone   Sergio Serrano
• [Asterisk-Users] avm c4, ptmp   ePyron Felix Deierlein
• [Asterisk-Users] CallPres screening DDI   ePyron Felix Deierlein
• [Asterisk-Users] Call Transfer Problems with Grandstream Budgetone 100 Phone   Steve Totaro
• [Asterisk-Users] Asterisk QOS working perfect using sveasoft 3.11g   Andrew Kohlsmith
• [Asterisk-Users] Asterisk QOS working perfect using sveasoft 3.11g   Andrew Kohlsmith
• [Asterisk-Users] VoIP experiences with Cable and DSL   Andrew Kohlsmith
• [Asterisk-Users] Re: IAX2 'no authority found' problem   Simon Ward
• [Asterisk-Users] Get MWI from Telco's voicemail   Andrew Kohlsmith
• [Asterisk-Users] Asterisk QOS working perfect using sveasoft 3.11g   Andrew Kohlsmith
• [Asterisk-Users] personal voicemail   Altus Snyman
• AW: [Asterisk-Users] Integrating an old PBX with Asterisk   Andrew Kohlsmith
• [Asterisk-Users] h323 gnugk to h323 asterisk and then to endpoint   Thomas Kuepper
• [Asterisk-Users] PRI/H323 gateway   Asmine Ouloube
• [Asterisk-Users] personal voicemail   Altus Snyman
• [Asterisk-Users] personal voicemail   Steve Totaro
• [Asterisk-Users] h323 gnugk to h323 asterisk and then to endpoint   eltorio
• [Asterisk-Users] personal voicemail   Dave Cotton
• [Asterisk-Users] Using answering machine in my phone   Adam Bezanson
• [Asterisk-Users] personal voicemail   Soren Rathje
• [Asterisk-Users] H323 Call Dropping   Sebastian Nocetti
• [Asterisk-Users] personal voicemail   Joe Dennick
• [Asterisk-Users] personal voicemail   Andrew Kohlsmith
• [Asterisk-Users] h323 gnugk to h323 asterisk and then to endpoint   Thomas Kuepper
• [Asterisk-Users] Using answering machine in my phone   Andrew Kohlsmith
• [Asterisk-Users] H323 Call Dropping   Thomas Kuepper
• [Asterisk-Users] h323 gnugk to h323 asterisk and then to endpoint   eltorio
• [Asterisk-Users] H323 Call Dropping   Asterisk .
• [Asterisk-Users] h323 gnugk to h323 asterisk and then to endpoint   eltorio
• [Asterisk-Users] No incoming audio on incoming SIP calls   John Howard
• [Asterisk-Users] Color in console   Steven Critchfield
• [Asterisk-Users] h323 gnugk to h323 asterisk and then to endpoint   Thomas Kuepper
• [Asterisk-Users] Cisco SIP Phone 7960 & DTMF Problem   Miroslav Nachev
• [Asterisk-Users] Cisco SIP Phone 7960 & DTMF Problem   Shane Young
• [Asterisk-Users] Asterisk does not disconnect SIP call   Thomas Heiss
• [Asterisk-Users] No incoming audio on incoming SIP calls   Thomas Kuepper
• [Asterisk-Users] shared voicemail   Altus Snyman
• [Asterisk-Users] PRI/H323 gateway   William Suffill
• [Asterisk-Users] First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs?   Mike Cathey
• [Asterisk-Users] H323 Call Dropping   Sebastian Nocetti
• [Asterisk-Users] X100P Kernel Panic   Tom Lawrence
• [Asterisk-Users] shared voicemail   William Suffill
• [Asterisk-Users] shared voicemail   Seth Remington
• [Asterisk-Users] Advice on possible set-up   Brent Franks
• [Asterisk-Users] No incoming audio on incoming SIP calls   John Howard
• [Asterisk-Users] Using answering machine in my phone   Adam Bezanson
• [Asterisk-Users] h323 gnugk to h323 asterisk and then to endpoint   Thomas Kuepper
• [Asterisk-Users] Avaya/Lucent Definity -> Asterisk interop question   Robert.Kelly at ebimed.com
• [Asterisk-Users] Bridging Calls   Jason R. Park
• [Asterisk-Users] Advice on possible set-up   Andrew Newton
• [Asterisk-Users] IAX2 'no authority found' problem   Tony Nichols
• [Asterisk-Users] Identifying which call an event belongs to   Michael Ulitskiy
• [Asterisk-Users] PRI protocol question...   John Harragin
• [Asterisk-Users] Using Cisco SIP Phones with Asterisk   Gary Carr
• [Asterisk-Users] Avaya/Lucent Definity -> Asterisk interop question   Scott Stingel
• [Asterisk-Users] PRI protocol question...   Scott Stingel
• [Asterisk-Users] problems with asterisk and the IAX protocol   Pamela Weis
• [Asterisk-Users] Avaya/Lucent Definity -> Asterisk interop question   Steve Kann
• [Asterisk-Users] Re: X100P Kernel Panic   Dave Hawkes
• [Asterisk-Users] Re: X100P Kernel Panic   Jason Stewart
• [Asterisk-Users] H323 Call Dropping   Jeremy McNamara
• [Asterisk-Users] PRI protocol question...   John Harragin
• [Asterisk-Users] asterisk+radius   Clayton
• [Asterisk-Users] Anyone use AdvancedVOIP ?   Darren Bentley
• [Asterisk-Users] transfering incoming message from app_queue   James Cloos
• [Asterisk-Users] DTMF after answer   John Bittner
• [Asterisk-Users] shared voicemail   Wayne
• [Asterisk-Users] new bounty for modifying calling card application to mysql   Sathya Weerasooriya
• [Asterisk-Users] new bounty for modifying calling card application to mysql   steve at daviesfam.org
• [Asterisk-Users] new bounty for modifying calling card application to mysql   Richard Cook
• [Asterisk-Users] shared voicemail   Seth Remington
• [Asterisk-Users] Anyone use AdvancedVOIP ?   Luke Catranis
• [Asterisk-Users] new bounty for modifying calling card application to mysql   Sebastian Nocetti
• [Asterisk-Users] PRI Errors... Ouch   Andrew McRory
• [Asterisk-Users] asterisk+radius   Lubomir Christov
• [Asterisk-Users] Avaya/Lucent Definity -> Asterisk interop question   kwijibo at zianet.com
• [Asterisk-Users] Asterisk used like SIP to h323 convertor   Narcis GRATIANU
• [Asterisk-Users] Using answering machine in my phone   Adam Bezanson
• [Asterisk-Users] Avaya/Lucent Definity -> Asterisk interop question   Ken Godee
• [Asterisk-Users] <<< MEETME_AGI_BACKGROUND inside MEET ME>>>   jeff quade
• [Asterisk-Users] Using answering machine in my phone   Roger Gulbranson
• [Asterisk-Users] Avaya/Lucent Definity -> Asterisk interop question   Mike Cathey
• [Asterisk-Users] Anyone use AdvancedVOIP ?   Kanuri, Seshu
• [Asterisk-Users] iConnectHere and CallerId   ERwin Hernandez
• [Asterisk-Users] shared voicemail   Wayne
• [Asterisk-Users] ASTRICON 2004   Rick Segrest
• [Asterisk-Users] Cisco SIP Phone 7960 & DTMF Problem   Randy Bush
• [Asterisk-Users] Re: <<< MEETME_AGI_BACKGROUND inside MEET ME>>>   Tony Mountifield
• [Asterisk-Users] iConnectHere and CallerId   Chris Shaw
• [Asterisk-Users] NAT problems   Bartosz Wegrzyn
• [Asterisk-Users] NAT problems   Chris Shaw
• [Asterisk-Users] voicemail attachment setup per user   Gary Carr
• [Asterisk-Users] voicemail attachment setup per user   Luke Catranis
• [Asterisk-Users] NAT problems   Bartosz Wegrzyn
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #4831 - 4 msgs   Andrew McRory
• [Asterisk-Users] NAT problems   Chris Shaw
• [Asterisk-Users] Strange message, and one-way audio between sip and H.323   Roberto Piola
• [Asterisk-Users] is asterisk and/or spandsp what I need for integration with Cisco?   Brian Jarrett
• [Asterisk-Users] NAT problems   Bartosz Wegrzyn
• [Asterisk-Users] NAT problems   Bartosz Wegrzyn
• [Asterisk-Users] NAT problems   Chris Shaw
• [Asterisk-Users] NAT problems   Chris Shaw
• [Asterisk-Users] Dialplan question   Simon Brown
• [Asterisk-Users] NAT problems   Bartosz Wegrzyn
• [Asterisk-Users] Skinny and CISCO 7905G   Alessandro Polverini
• [Asterisk-Users] NAT problems   Chris Shaw
• [Asterisk-Users] Re: Iaxy issue   Glen Hinkle
• [Asterisk-Users] asterisk+radius   Duane Cox
• [Asterisk-Users] users   Travis Conway
• [Asterisk-Users] NAT problems   Bartosz Wegrzyn
• [Asterisk-Users] Skinny and CISCO 7905G   Kubat, Philip
• [Asterisk-Users] users   Brian Roy
• [Asterisk-Users] is asterisk and/or spandsp what I need for integration with Cisco?   Lee Howard
• [Asterisk-Users] iptables, Cisco 7960 and TFTP   Chad Brown
• [Asterisk-Users] Sip dialback   Steve Szmidt
• [Asterisk-Users] Dialplan question   Holger Schurig
• [Asterisk-Users] Dialplan question   steve at daviesfam.org
• [Asterisk-Users] iptables, Cisco 7960 and TFTP   Michael Løjtnant
• [Asterisk-Users] Dialplan question   Florian Overkamp
• [Asterisk-Users] Urgent help with Sip <------> H323 on FREEBSD   Krystian Filiks
• [Asterisk-Users] tdm400p lockups   Ryan Courtnage
• [Asterisk-Users] Re: Invalid data ( 4 bytes at the end) message - g729codec   Yap Teong Eng
• [Asterisk-Users] Microsoft RTC Authentication   blas frances
• [Asterisk-Users] First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs?   Jason Williams
• [Asterisk-Users] Entry point & Minimal requirements for Linux Device Driver and connection with Asterisk IP PBX   Miroslav Nachev
• [Asterisk-Users] iax->asterisk->capi isdn; crackle overlay after about 1 minute   cdeipenbrock at gmx.de
• [Asterisk-Users] tdm400p lockups   Ryan Courtnage
• [Asterisk-Users] No incoming audio on incoming SIP calls   Jason Williams
• [Asterisk-Users] ZAPTEL & ZAPATA   Miroslav Nachev
• [Asterisk-Users] tdm400p lockups   Ryan Courtnage
• [Asterisk-Users] No incoming audio on incoming SIP calls   John Howard
• [Asterisk-Users] Lots of Echo with SIP -> Asterisk -> PSTN   James Freire
• [Asterisk-Users] DTMF after answer   Brent Franks
• [Asterisk-Users] RE: Zaptel X100P Kernel Panic   Tom Lawrence
• [Asterisk-Users] Paging with Sipura 3000 FXO   Paul Zimm
• [Asterisk-Users] DTMF after answer   Marc C Storck
• [Asterisk-Users] <<< MEETME_AGI_BACKGROUND inside MEET ME>>>   jeff quade
• [Asterisk-Users] Re: Iaxy issue   Justin Carlson
• [Asterisk-Users] DTMF after answer   Marc C Storck
• [Asterisk-Users] After RC1 upgrade, temporary loss of voice   Adams, Gavin-ML
• [Asterisk-Users] Re: <<< MEETME_AGI_BACKGROUND inside MEET ME>>>   Tony Mountifield
• [Asterisk-Users] Zaptel CVS HEAD 08-05-2004 problems   Ryan Courtnage
• [Asterisk-Users] Difficulty evaluating the return value of PlayBack (or any other extensions.conf command   Andreas Sikkema
• [Asterisk-Users] DTMF after answer   Nicolas Gudino
• [Asterisk-Users] Avaya/Lucent Definity -> Asterisk interop question   Steve Kann
• [Asterisk-Users] MPG123, Music On Hold and Variable Bit Rate   Chris Stenton
• [Asterisk-Users] E1 monochannel :-(   Horacio J. Peña
• [Asterisk-Users] tdm400p lockups   Tim Sailer
• [Asterisk-Users] H323 Call Dropping   Asterisk .
• [Asterisk-Users] E1 monochannel :-(   Nicolas Gudino
• [Asterisk-Users] Asterisk as SIP proxy?   Terry Wilson
• [Asterisk-Users] tdm400p lockups   Joshua M. Thompson
• [Asterisk-Users] E1 monochannel :-(   Marc C Storck
• [Asterisk-Users] Sip dialback   Steve Szmidt
• [Asterisk-Users] hang up when going to voicemail   awesome at surfbest.net
• [Asterisk-Users] VoicemailMain Issues   Robert Jackson
• [Asterisk-Users] First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs?   Mike Cathey
• [Asterisk-Users] tdm400p lockups   Joshua M. Thompson
• [Asterisk-Users] Lots of Echo with SIP -> Asterisk -> PSTN   Chad Brown
• [Asterisk-Users] E1 monochannel :-(   steve at daviesfam.org
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   Sebastian Nocetti
• [Asterisk-Users] Lots of Echo with SIP -> Asterisk -> PSTN   James Freire
• [Asterisk-Users] tdm400p lockups   Tim Sailer
• [Asterisk-Users] Asterisk not starting   Andreas Roedl
• [Asterisk-Users] Asterisk not starting   Jeremy McNamara
• [Asterisk-Users] VoicemailMain Issues   Seth Remington
• [Asterisk-Users] Inbound not working with iconnect   Raj
• [Asterisk-Users] Anyone use AdvancedVOIP ?   Clayton
• [Asterisk-Users] FXO Problems   Mike Coakley
• [Asterisk-Users] MPG123, Music On Hold and Variable Bit Rate   Mike Coakley
• [Asterisk-Users] tdm400p lockups   Hez R
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   mattf
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   Sebastian Nocetti
• [Asterisk-Users] meetme   Travis Conway
• [Asterisk-Users] hang up when going to voicemail   awesome at surfbest.net
• [Asterisk-Users] Difficulty evaluating the return value of PlayBack (or any other extensions.conf command   Philipp von Klitzing
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   mattf
• [Asterisk-Users] meetme   Joshua Colp
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   Sebastian Nocetti
• [Asterisk-Users] Re: meetme   Tony Mountifield
• [Asterisk-Users] Zaptel CVS HEAD 08-05-2004 problems   Mike Coakley
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   mattf
• [Asterisk-Users] ShoreTel Phones   Joseph Finley
• [Asterisk-Users] Zaptel CVS HEAD 08-05-2004 problems   Mike Coakley
• [Asterisk-Users] t100 error on FC2   Jason Kawakami
• [Asterisk-Users] Re: meetme   Travis Conway
• [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)   Andrew Kohlsmith
• [Asterisk-Users] Interesting catalog: Viking Electronics   John Todd
• [Asterisk-Users] After RC1 upgrade, temporary loss of voice   Andrew Kohlsmith
• [Asterisk-Users] FXO Problems   Mike Coakley
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   Robert Hanzlik
• [Asterisk-Users] Zaptel CVS HEAD 08-05-2004 problems   Steve Szmidt
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   Robert Hajime Lanning
• [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)   steve at daviesfam.org
• [Asterisk-Users] After RC1 upgrade, temporary loss of voice   steve at daviesfam.org
• [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)   Andrew Kohlsmith
• [Asterisk-Users] Interesting catalog: Viking Electronics   mattf
• [Asterisk-Users] After RC1 upgrade, temporary loss of voice   Andrew Kohlsmith
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   Jorge Mendoza
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   William Boehlke
• [Asterisk-Users] no mail sent on voice message   Dan Fernandez
• [Asterisk-Users] termcapsupport not found   dirk los
• [Asterisk-Users] H323 Call Dropping   Asterisk .
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   Scott Stingel
• [Asterisk-Users] redhat 9 and oh323   mohammad mirzaee
• [Asterisk-Users] Problems loading chan_h323 on Opteron 64 bit   Roger Schreiter
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   Andres
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   Roger Schreiter
• [Asterisk-Users] H323 Call Dropping   Tenorio, Leandro
• [Asterisk-Users] tdm400p lockups   Greg Boehnlein
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   Scott Stingel
• [Asterisk-Users] Problems loading chan_h323 on Opteron 64 bit   Wolfgang S. Rupprecht
• [Asterisk-Users] Need help on installing an Asterisk PBX   Wayne Xie
• [Asterisk-Users] H323 Call Dropping   Asterisk .
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   Horacio J. Peña
• [Asterisk-Users] Asterisk Dry Run   niko singh
• [Asterisk-Users] ASTERISK AND 120 CONCURRENT CALLS   Greg Boehnlein
• [Asterisk-Users] tdm400p lockups   Joshua M. Thompson
• [Asterisk-Users] First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs?   Leo Ann Boon
• [Asterisk-Users] oem x100p undefined symbol ast_get_txt   Lyle Giese
• [Asterisk-Users] oem x100p undefined symbol ast_get_txt   William Suffill
• [Asterisk-Users] oem x100p undefined symbol ast_get_txt   Lyle Giese
• [Asterisk-Users] Asterisk and Cisco Call Manager.   Gurdeep Singh Bagga Guru
• [Asterisk-Users] Interesting catalog: Viking Electronics   David Hickman
• [Asterisk-Users] NAT problems attn Chris Shaw   Bartosz Wegrzyn
• [Asterisk-Users] Asterisk, channel banks and SDH   Flynn
• [Asterisk-Users] Asterisk Dry Run   Greg Hill
• [Asterisk-Users] Vonage working with asterisk   Assaf Benharoosh
• [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)   Andrew Kohlsmith
• [Asterisk-Users] iaxtel, asterisk, and sipura 1000 am having trouble with codecs   hank
• [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)   Andrew Kohlsmith
• [Asterisk-Users] Vonage working with asterisk   hank
• [Asterisk-Users] Interesting catalog: Viking Electronics   Jay Milk
• [Asterisk-Users] Vonage working with asterisk   Jay Milk
• [Asterisk-Users] Interesting catalog: Viking Electronics   Wolfgang S. Rupprecht
• [Asterisk-Users] Asterisk Dry Run   Dave Cotton
• [Asterisk-Users] Asterisk : No Sound Issues   niko singh
• [Asterisk-Users] Confused --> Hardware specs   Asterisk
• [Asterisk-Users] Re: Invalid data ( 4 bytes at the end) message - from g729codec   teyap at nextwavecomm.com.my
• [Asterisk-Users] Asterisk and Douglas Telecom   Roland Zagler
• [Asterisk-Users] res_config_odbc not working   Dennis Nacino
• [Asterisk-Users] redhat 9 and oh323   joachim
• [Asterisk-Users] Video Calls between SIP and H.323   Bernie Hoeneisen
• TEMPORARY FIX: Re: [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)   Andrew Kohlsmith
• [Asterisk-Users] After RC1 upgrade, temporary loss of voice   Andrew Kohlsmith
• [Asterisk-Users] (no subject)   Marc C Storck
• [Asterisk-Users] Howto do something after a Dial command is Answered?   nospams
• [Asterisk-Users] Howto do something after a Dial command is Answered?   nospams
• [Asterisk-Users] Howto do something after a Dial command is Answered?   Kannaiyan Natesan
• [Asterisk-Users] Howto do something after a Dial command is Answered?   nospams
• [Asterisk-Users] astcc help   Sathya Weerasooriya
• [Asterisk-Users] Howto do something after a Dial command is Answered?   Asterisk .
• [Asterisk-Users] Howto do something after a Dial command is Answered?   nospams
• [Asterisk-Users] Inbound not working with iconnect   Greg Blakely
• [Asterisk-Users] Vonage working with asterisk   Greg Blakely
• [Asterisk-Users] Inbound not working with iconnect   Raj
• [Asterisk-Users] Howto do something after a Dial command is Answered?   Seth Remington
• [Asterisk-Users] Howto do something after a Dial command is Answered?   nospams
• [Asterisk-Users] Inbound not working with iconnect   Luke Catranis
• [Asterisk-Users] voicemail attachment setup per user   Seth Remington
• [Asterisk-Users] Inbound not working with iconnect   Greg Blakely
• [Asterisk-Users] Inbound not working with iconnect   Raj
• [Asterisk-Users] Inbound not working with iconnect   Raj
• [Asterisk-Users] Inbound not working with iconnect   Luke Catranis
• [Asterisk-Users] Asterisk : No Sound Issues   Greg Hill
• [Asterisk-Users] call for help from uip200 users   Ryan Courtnage
• [Asterisk-Users] Lots of FXS ports / Channel Bank ?   Joel Vandal
• [Asterisk-Users] System Requirements   Beierlein Moritz
• [Asterisk-Users] Getting a modem connection working via Asterisk   Nate Carlson
• [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)   hank
• [Asterisk-Users] Interesting catalog: Viking Electronics   hank
• [Asterisk-Users] Asterisk : No Sound Issues   hank
• [Asterisk-Users] Lots of FXS ports / Channel Bank ?   Shane Young
• [Asterisk-Users] Message waiting   Don Hughes
• [Asterisk-Users] Message waiting   William Suffill
• [Asterisk-Users] System Requirements   William Suffill
• [Asterisk-Users] System Requirements   Beierlein Moritz
• [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)   Andrew Kohlsmith
• [Asterisk-Users] Asterisk not starting   Andreas Roedl
• [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)   T. Chan
• [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)   T. Chan
• [Asterisk-Users] Where to get latest SIP bootrom/firmware for Polycom IP500 phones?   Lenny Tropiano / asterisk.org Mailing list
• [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio) (fwd)   steve at daviesfam.org
• [Asterisk-Users] H.323 IVR with avaya multivantage.   jong suk Lee
• [Asterisk-Users] Mediatrix 1204, where I can get the last firmware and mib file ?   miguel at amplanet.com.br
• [Asterisk-Users] Mediatrix 1204 - Error: Operation not permitted   miguel at amplanet.com.br
• [Asterisk-Users] Generic X100P setup issues   Graham W. Mitchell
• [Asterisk-Users] Mediatrix 1204 - Error: Operation not permitted   Bob Knight
• [Asterisk-Users] Asterisk : No Sound No Dial   niko singh
• [Asterisk-Users] Generic X100P setup issues   Philip Fleischer
• [Asterisk-Users] OH323 Call Dropping [was: H323 Call Dropping]   Asterisk .
• [Asterisk-Users] oem x100p undefined symbol ast_get_txt   Lyle Giese
• [Asterisk-Users] OH323 Call Dropping [was: H323 Call Dropping]   jong suk Lee
• [Asterisk-Users] voice mail greeting not updating?   hank
• [Asterisk-Users] Generic X100P setup issues   hank
• [Asterisk-Users] WARNING[1264581056]   Sudhir Kumar
• [Asterisk-Users] WARNING[1264581056]   Jeremy McNamara
• [Asterisk-Users] SOLVED: 100% cpu usage causes big problems   Tracy R Reed
• [Asterisk-Users] System Reqirements HELP   Beierlein Moritz
• [Asterisk-Users] System Reqirements HELP   Florian Overkamp
• [Asterisk-Users] System Reqirements HELP   Steven P. Donegan
• [Asterisk-Users] No Sound and Jungle:   niko singh
• [Asterisk-Users] Message waiting   Reid A. Forrest
• [Asterisk-Users] Mediatrix 1204, where I can get the last firmware and mib file ?   Rich Adamson
• [Asterisk-Users] Generic X100P setup issues   Lyle Giese
• [Asterisk-Users] Generic X100P setup issues   Ed Maste
• [Asterisk-Users] No Sound and Jungle:   Dave Cotton
• [Asterisk-Users] Asterisk : No Sound No Dial   Mike Coakley
• [Asterisk-Users] Asterisk : No Sound No Dial   Greg Hill
• [Asterisk-Users] Voicepulse problems?   Bruce Komito
• [Asterisk-Users] System Reqirements HELP   Jay Milk
• [Asterisk-Users] Voicepulse problems?   Ken Wiesner
• [Asterisk-Users] Voicepulse problems?   Greg Blakely
• [Asterisk-Users] Voicepulse problems?   dialtone9 at phreaker.net
• [Asterisk-Users] Voicepulse problems?   Steve Totaro
• [Asterisk-Users] System Reqirements HELP   Steven P. Donegan
• [Asterisk-Users] Voicepulse problems?   Steve Totaro
• [Asterisk-Users] Asterisk not starting   Andreas Roedl
• [Asterisk-Users] Asterisk not starting - SOLVED!   Andreas Roedl
• [Asterisk-Users] Asterisk not starting - SOLVED!   Dave Cotton
• [Asterisk-Users] Asterisk not starting - SOLVED!   Andreas Roedl
• [Asterisk-Users] Voicepulse problems?   Greg Blakely
• [Asterisk-Users] Asterisk not starting - SOLVED!   Steven Critchfield
• [Asterisk-Users] truncated extensions   Kevin Johnson
• [Asterisk-Users] Zaptel.conf and Zapata.conf for TDM12B   Shawn Lipman
• [Asterisk-Users] Voicepulse problems?   Steve Totaro
• [Asterisk-Users] asterisk-update script   Steve Szmidt
• [Asterisk-Users] asterisk-update script - and the script   Steve Szmidt
• [Asterisk-Users] asterisk-update script - and the script - Fixed typo   Steve Szmidt
• [Asterisk-Users] How to notify the user about new message using SMS   Bartosz Wegrzyn
• [Asterisk-Users] System Requirements   Christoph Rothe
• [Asterisk-Users] Re: instable Modem-Module in CVS ?   Christoph Rothe
• [Asterisk-Users] stable-RC1   administrator tootai
• [Asterisk-Users] System Requirements   Rudy.Prakash at peregrine-semi.com.au
• [Asterisk-Users] asterisk-update script - and the script - Fixed typo   Steve Szmidt
• [Asterisk-Users] System Requirements   Rudy.Prakash at peregrine-semi.com.au
• [Asterisk-Users] How to notify the user about new message using SMS   Steven Critchfield
• [Asterisk-Users] How to notify the user about new message using SMS   usedcanon
• [Asterisk-Users] System Requirements   Steven Critchfield
• [Asterisk-Users] Generic X100P setup issues   hank
• [Asterisk-Users] System Reqirements HELP   hank
• [Asterisk-Users] Voicepulse problems?   hank
• [Asterisk-Users] asterisk-update script   hank
• [Asterisk-Users] asterisk-update script   William Suffill
• [Asterisk-Users] Douglas Telecom & Asterisk   Roland Zagler
• [Asterisk-Users] Asterisk not starting - SOLVED!   Nicholas Bachmann
• [Asterisk-Users] System Requirements   Ed Maste
• [Asterisk-Users] truncated extensions   Greg Hill
• [Asterisk-Users] System Reqirements HELP   Jay Milk
• [Asterisk-Users] Generic X100P setup issues   Graham W. Mitchell
• [Asterisk-Users] pbx answers after answering from analog phone   Joe Dennick
• [Asterisk-Users] pbx answers after answering from analog phone   Lyle Giese
• [Asterisk-Users] pbx answers after answering from analog phone   Lyle Giese
• [Asterisk-Users] asterisk-update script   hank
• [Asterisk-Users] iconnect inbound - so do we know how to fix it   Sathya Weerasooriya
• [Asterisk-Users] Queue issues   ghost at silcon.com
• [Asterisk-Users] Straight to VoiceMail   Simon Brown
• [Asterisk-Users] Extension Status on different phone   Adam Goryachev
• [Asterisk-Users] Howto configure TE410P card and channels   Snak Newyork
• [Asterisk-Users] How to notify the user about new message using SMS   Florian Overkamp
• [Asterisk-Users] pbx answers after answering from analog phone   Trevor Peirce
• [Asterisk-Users] Difficulty evaluating the return value of PlayBack (or any other extensions.conf command   Andreas Sikkema
• [Asterisk-Users] iconnect inbound - so do we know how to fix it   Vladyslav
• FW: [Asterisk-Users] problems with asterisk and the IAX protocol   Pamela Weis
• [Asterisk-Users] res_config_odbc not working   Dennis Nacino
• [Asterisk-Users] e164.lu   Marc C Storck
• [Asterisk-Users] traffic termination around the globe?   Roy Sigurd Karlsbakk
• [Asterisk-users] Some Error on Asterisk....   Nilesh sonavani
• [Asterisk-users] Some Error on Asterisk....   ShanKutti
• [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point & DDI   Nick Barnes
• [Asterisk-Users] Sound file quality   David Gurr
• [Asterisk-Users] Howto configure TE410P card and channels   Steven Critchfield
• [Asterisk-Users] Sound file quality   Steven Critchfield
• [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point & DDI   Michael Sandee
• [Asterisk-Users] h323 direkt call instead over GK   Thomas Kuepper
• [Asterisk-Users] Variables In a Context   Simon
• [Asterisk-Users] Some Errors on Asterisk   Nilesh sonavani
• [Asterisk-Users] Some Errors on Asterisk server   Nilesh sonavani
• [Asterisk-Users] Which is PCI HDLC Chip used for TE405P / TE410P / E100P ?   Miroslav Nachev
• [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point & DDI   Holger Schurig
• [Asterisk-Users] Inbound not working with iconnect   Raj
• [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point & DDI   Nick Barnes
• [Asterisk-Users] iconnect inbound - so do we know how to fix it   Raj
• [Asterisk-Users] pbx answers after answering from analog phone   Lyle Giese
• [Asterisk-Users] Re: Sound file quality   Tony Mountifield
• [Asterisk-Users] How do folks handle NAT routing?   David Gurr
• [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point & DDI   Maurizio Marini
• [Asterisk-Users] sip endpoint not ringing   Thomas Kuepper
• [Asterisk-Users] asterisk-update script   Leif Madsen
• [Asterisk-Users] RC1 - callparking   administrator tootai
• [Asterisk-Users] asterisk with H.323 phone   ml_asterisk-users at rajnoch.sk
• [Asterisk-Users] Click to Call   Andrei Goncalves
• [Asterisk-Users] termcapsupport not found   Kevin Walsh
• [Asterisk-Users] Asterisk not starting - SOLVED!   Andreas Roedl
• [Asterisk-Users] asterisk-update script   Steve Szmidt
• [Asterisk-Users] iconnect inbound - so do we know how to fix it   Greg Blakely
• [Asterisk-Users] ring tone   Thomas Kuepper
• [Asterisk-Users] How to notify the user about new message using SMS   Bartosz Wegrzyn
• [Asterisk-Users] How to notify the user about new message using SMS   Bartosz Wegrzyn
• [Asterisk-Users] OH323 Call Dropping [was: H323 Call Dropping]   Asterisk .
• [Asterisk-Users] New feature request   Bart Coppens
• [Asterisk-Users] uniden phones   Gary Carr
• [Asterisk-Users] ChangeMonitor syntax   eduardo at garnier.inf.br
• [Asterisk-Users] iconnect inbound - so do we know how to fix it   Andrew Thompson
• [Asterisk-Users] Click to Call   Andrew Thompson
• [Asterisk-Users] termcapsupport not found   Steven Critchfield
• [Asterisk-Users] ChangeMonitor syntax   Seth Remington
• [Asterisk-Users] Questionaire :   niko singh
• [Asterisk-Users] uniden phones   James H. Thompson
• [Asterisk-Users] Questionaire :   Joseph
• [Asterisk-Users] Questionaire :   Aleph Communications
• [Asterisk-Users] Call File Routing   Aleph Communications
• [Asterisk-Users] MUSIC ON HOLD PLAYING like in slow motion   Bartosz Wegrzyn
• [Asterisk-Users] ChangeMonitor syntax   Holger Schurig
• [Asterisk-Users] MUSIC ON HOLD PLAYING -SOLVED   Bartosz Wegrzyn
• [Asterisk-Users] How do folks handle NAT routing?   Andres
• [Asterisk-Users] uniden phones   Paul Zimm
• [Asterisk-Users] Sound file quality   hank
• [Asterisk-Users] Click to Call   Glen Hinkle
• [Asterisk-Users] Click to Call   Andrew Thompson
• [Asterisk-Users] Asterisk and Cisco Call Manager.   Andres Junge
• [Asterisk-Users] ChangeMonitor syntax   Seth Remington
• [Asterisk-Users] Click to Call   Steven Critchfield
• [Asterisk-Users] Strange H323 problem   Andres Junge
• [Asterisk-Users] Call File Routing   Jeremy Hall
• [Asterisk-Users] iconnect inbound - so do we know how to fix it   Sathya Weerasooriya
• [Asterisk-Users] Click to Call   Brian Capouch
• [Asterisk-Users] Fedora FC2 and Zaptel (Torisa)   Matt Schulte
• [Asterisk-Users] inbound/outbound trunk groups   Ravi Hanumaiah
• [Asterisk-Users] Re: Grandstream Message Waiting light   Stephen R. Besch
• [Asterisk-Users] Click to Call   Andrew Thompson
• [Asterisk-Users] Fedora FC2 and Zaptel (Torisa)   Oliver
• [Asterisk-Users] Re: Sound file quality   Tony Mountifield
• [Asterisk-Users] Call File Routing   Aleph Communications
• [Asterisk-Users] uniden phones   Nathan C. Smith
• [Asterisk-Users] Sound file quality   Rick L. Wilson, Sr.
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Christopher L. Wade
• [Asterisk-Users] Fedora FC2 and Zaptel (Torisa)   Matt Schulte
• [Asterisk-Users] quadBRI + FAX   irmantas.gudelis at vilnius2.balt.net
• [Asterisk-Users] Click to Call   Ed Guy
• [Asterisk-Users] Re: Grandstream Message Waiting light   Chris Shaw
• [Asterisk-Users] Fedora FC2 and Zaptel (Torisa)   Oliver
• [Asterisk-Users] H323 under asterisk RC1 ?   Roberto Piola
• [Asterisk-Users] quadBRI + FAX   Michael Sandee
• [Asterisk-Users] called and callers buttons on bt100   Jason Kawakami
• [Asterisk-Users] Application asterisk uses obsolete OSS audio interface   Travis Conway
• [Asterisk-Users] Snom Intercom   rayers.list
• [Asterisk-Users] Inbound Call Errors...   Stephen Malenshek
• [Asterisk-Users] h323 direkt call instead over GK   Jeremy McNamara
• [Asterisk-Users] CVS download   Travis Conway
• [Asterisk-Users] Snom Intercom   Steve Szmidt
• [Asterisk-Users] Fedora FC2 and Zaptel (Torisa)   Jean-Yves Avenard
• [Asterisk-Users] Questionaire :   matt.riddell at sineapps.com
• [Asterisk-Users] Fedora FC2 and Zaptel (Torisa)   asterisk at bsius.com
• [Asterisk-Users] Sound file quality   James Cloos
• [Asterisk-Users] Snom Intercom   Ryan Ayers
• [Asterisk-Users] Answer Call Waiting from Call Forward to Cell Phone   Matt Fontaine
• [Asterisk-Users] Snom Intercom   Michael Welter
• [Asterisk-Users] Answer Call Waiting from Call Forward to Cell Phone   Steve Szmidt
• [Asterisk-Users] Snom Intercom   Steve Szmidt
• [Asterisk-Users] asterisk with H.323 phone   Manoj Kr. Gupta
• [Asterisk-Users] introduced Agents and * stops answering calls   Sam Tilders
• [Asterisk-Users] Click to Call   VoIP
• [Asterisk-Users] truncated extensions   Kevin Johnson
• [Asterisk-Users] 831 Santa Cruz/Watsoncille, Calif. DIDs   lists-jmhunter
• [Asterisk-Users] 831 Santa Cruz/Watsoncille, Calif. DIDs   Luke Catranis
• [Asterisk-Users] 831 Santa Cruz/Watsoncille, Calif. DIDs   lists-jmhunter
• [Asterisk-Users] 831 Santa Cruz/Watsoncille, Calif. DIDs   Luke Catranis
• [Asterisk-Users] TE410P-RED Alarm   SipMonster
• [Asterisk-Users] ChangeMonitor syntax   Holger Schurig
• [Asterisk-Users] Re: CVS download   Tony Mountifield
• [Asterisk-Users] 831 Santa Cruz/Watsoncille, Calif. DIDs   hank
• [Asterisk-Users] AVM B1, chan_capi, Kernel 2.6   Stefan Tichy
• [Asterisk-Users] h.323 channel problem: I hear nothing   ml_asterisk-users at rajnoch.sk
• [Asterisk-Users] Problem with EuroISDN E1   Claus Futtrup
• [Asterisk-Users] [Asterisk-Users]help me in voice jittering problem   Atif Azhar
• [Asterisk-Users] Problem with EuroISDN E1   Storer, Darren
• [Asterisk-Users] Problem with EuroISDN E1   Claus Futtrup
• [Asterisk-Users] CAPI call transfer   Jonathan
• [Asterisk-Users] CAPI call transfer   Roland Zagler
• [Asterisk-Users] just a few newbie questions   Chris Lee
• [Asterisk-Users] Problem with EuroISDN E1   Antonio Rabena
• [Asterisk-Users] just a few newbie questions   asterisk at bortal.de
• [Asterisk-Users] CAPI call transfer   Jonathan
• [Asterisk-Users] just a few newbie questions   Holger Schurig
• [Asterisk-Users] CAPI call transfer   Roland Zagler
• [Asterisk-Users] Problem with EuroISDN E1   Storer, Darren
• [Asterisk-Users] CAPI call transfer   Jonathan
• [Asterisk-Users] just a few newbie questions   asterisk at bortal.de
• [Asterisk-Users] Firefly and *... Argh!   David MacKinnon
• [Asterisk-Users] RTP-problems with voipexchange.ru   Roger Schreiter
• [Asterisk-Users] CAPI call transfer   Roland Zagler
• [Asterisk-Users] Sound problems ..linux red hat   niko singh
• [Asterisk-Users] Problem with EuroISDN E1   Claus Futtrup
• [Asterisk-Users] just a few newbie questions   Holger Schurig
• [Asterisk-Users] Firefly and *... Argh!   Robert Barnes
• [Asterisk-Users] Sjphone Troubles :   niko singh
• [Asterisk-Users] Firefly and *... Argh!   David MacKinnon
• [Asterisk-Users] Sound file quality   Christoph Rothe
• [Asterisk-Users] Firefly and *... Argh!   Robert Barnes
• [Asterisk-Users] CAPI call transfer   Jonathan
• [Asterisk-Users] Firefly and *... Argh!   David MacKinnon
• [Asterisk-Users] CAPI call transfer   Roland Zagler
• [Asterisk-Users] CAPI call transfer   Jonathan
• [Asterisk-Users] Click to Call   Ed Guy
• [Asterisk-Users] CAPI call transfer   Roland Zagler
• [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point & DDI   stan at saticed.me.uk
• [Asterisk-Users] asterisk mirror   asterisk at bortal.de
• [Asterisk-Users] CAPI call transfer   Jonathan
• [Asterisk-Users] called and callers buttons on bt100   Seth Remington
• [Asterisk-Users] Fedora FC2 and Zaptel (Torisa)   Matt Schulte
• [Asterisk-Users] just a few newbie questions   Lee Allen
• [Asterisk-Users] asterisk mirror   Steven Critchfield
• [Asterisk-Users] asterisk mirror   asterisk at bortal.de
• [Asterisk-Users] T400P modprobe/ztcfg failure   Matt Schulte
• [Asterisk-Users] Config for Asterisk with Sipgate behind Linksys Router   box100
• [Asterisk-Users] iconnect inbound - so do we know how to fix it   Paul Cheng
• [Asterisk-Users] distinguish rining tone   Bartosz Jozwiak
• [Asterisk-Users] TxtCIDName returning mangled text   Jeff Pyle
• [Asterisk-Users] [OT]Google and the Asterisk list   Michael Welter
• [Asterisk-Users] WiFi phone radiation regulation?   Leo Ann Boon
• [Asterisk-Users] Can Incomming CallerID be fowarded to a SIP phone extension?   James Freire
• [Asterisk-Users] just a few newbie questions   Holger Schurig
• [Asterisk-Users] TE410P-RED Alarm   Scott Stingel
• [Asterisk-Users] asterisk mirror   Seth Remington
• [Asterisk-Users] truncated extensions   Greg Hill
• [Asterisk-Users] [OT]Google and the Asterisk list   Seth Remington
• [Asterisk-Users] DTMF issues   AJ Grinnell
• [Asterisk-Users] Asterisk in a DMZ   Bastian Schern
• [Asterisk-Users] asterisk mirror   William Suffill
• [Asterisk-Users] HowTo test asterisk in internal network?   asterisk at bortal.de
• [Asterisk-Users] H323 gateway   Asmine Ouloube
• [Asterisk-Users] Pulver wisip firmware?   asterisk at bsius.com
• [Asterisk-Users] called and callers buttons on bt100   Steve Szmidt
• [Asterisk-Users] Asterisk in a DMZ   Joseph
• [Asterisk-Users] agent login   Greg Smith
• [Asterisk-Users] Asterisk in a DMZ   Andres Tello Abrego
• [Asterisk-Users] WiFi phone radiation regulation?   Dominique Kull
• [Asterisk-Users] Compile error H323   AsteriskList
• [Asterisk-Users] agent login   Peter Svensson
• [Asterisk-Users] Asterisk in a DMZ   Dominique Kull
• [Asterisk-Users] agent login   Robert Jackson
• [Asterisk-Users] Blocking the 'Do Not Call" List   drodden
• [Asterisk-Users] Blocking the 'Do Not Call" List   Chris Shaw
• [Asterisk-Users] Compile error H323   Jeremy McNamara
• [Asterisk-Users] Intriguing * problem with voicemail signalling   Richard Fall
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Christopher L. Wade
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Chris Shaw
• [Asterisk-Users] Sound file quality   James Cloos
• [Asterisk-Users] Blocking the 'Do Not Call" List   mattf
• [Asterisk-Users] Inbound Call Errors...   Stephen Malenshek
• [Asterisk-Users] called and callers buttons on bt100   Brian Capouch
• [Asterisk-Users] Blocking the 'Do Not Call" List   Josh Roberson
• [Asterisk-Users] DTMF issues   Greg Hill
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Josh Roberson
• [Asterisk-Users] Answer Call Waiting from Call Forward to Cell Phone   Matt Fontaine
• [Asterisk-Users] Pulver wisip firmware?   Ed Guy
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Christopher L. Wade
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Seth Remington
• [Asterisk-Users] Asterisk and Cisco Call Manager.   Chad Whitten
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Christopher L. Wade
• [Asterisk-Users] CAPI call transfer   Roland Zagler
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   John Todd
• [Asterisk-Users] DTMF issues   AJ Grinnell
• [Asterisk-Users] Blocking the 'Do Not Call" List   drodden
• [Asterisk-Users] iconnect inbound - FIXED (kinda)   Greg Blakely
• [Asterisk-Users] Asterisk in a DMZ   Steve Szmidt
• [Asterisk-Users] Blocking the 'Do Not Call" List   Chris Shaw
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Chris Shaw
• [Asterisk-Users] Blocking the 'Do Not Call" List   Chris Shaw
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   hank
• [Asterisk-Users] HowTo test asterisk in internal network?   Jay Milk
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Christopher L. Wade
• [Asterisk-Users] DTMF issues   Greg Hill
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Christopher L. Wade
• [Asterisk-Users] Blocking the 'Do Not Call" List   mattf
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   Nate Carlson
• [Asterisk-Users] Asterisk and Cisco Call Manager.   Andres Junge
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Sean Cheesman
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   Michael Welter
• [Asterisk-Users] Pulver wisip firmware?   Tim McKee
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   Soren Rathje
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   Loek Gijben
• [Asterisk-Users] Problem with EuroISDN E1   Storer, Darren
• [Asterisk-Users] Re: CVS download   Travis Conway
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Christopher L. Wade
• [Asterisk-Users] Polycom IP 500 - MWI Not Working   Wiley E. Siler
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   Nate Carlson
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   Wolfgang S. Rupprecht
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   Scott Laird
• [Asterisk-Users] Polycom IP 500 - MWI Not Working   Robert Jackson
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   Michael Welter
• [Asterisk-Users] Fedora FC2 and Zaptel (Torisa)   Oliver
• [Asterisk-Users] SNOM 200 and Asterisk Woes   Dan Mahoney, System Admin
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   james edwards
• [Asterisk-Users] Polycom IP 500 - MWI Not Working   niles at atheos.net
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Christopher L. Wade
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   cveazey at blackhillsfiber.com
• [Asterisk-Users] Parked Call Viewer Application - Freeware   Kyle Hagan
• [Asterisk-Users] CVS version tags   Ryan Parlee
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   Nate Carlson
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   Nate Carlson
• [Asterisk-Users] Polycom IP 500 - MWI Not Working   Wiley E. Siler
• [Asterisk-Users] UK SMS troubles   Darren Poulson
• [Asterisk-Users] Kernel 2.6 and zaptel data   asterisk-users at ics-il.net
• [Asterisk-Users] SNOM 200 and Asterisk Woes   Todd Lieberman
• [Asterisk-Users] SIP Transfers (Possibly reinvite)   Christopher Jacob
• [Asterisk-Users] SIP Transfers (Possibly reinvite)   Chris Shaw
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   Walt Reed
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   Chris Shaw
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   Scott Laird
• [Asterisk-Users] Kernel 2.6 and zaptel data   Kevin Walsh
• [Asterisk-Users] CVS version tags   Robert Jackson
• [Asterisk-Users] Problems with E100P   Carlos Hernandez
• [Asterisk-Users] truncated extensions   Kevin Johnson
• [Asterisk-Users] Fedora FC2 and Zaptel (Torisa)   Jean-Yves Avenard
• [Asterisk-Users] Kernel 2.6 and zaptel data   Jean-Yves Avenard
• [Asterisk-Users] Personal Meetme conferences; is there a better way to do this?   Stephen Uhler
• [Asterisk-Users] Re: [Asterisk-Dev] VoIP SPAM, what's next ?   Duane
• [Asterisk-Users] Blocking the 'Do Not Call" List   Steve Szmidt
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   hank
• [Asterisk-Users] Blocking the 'Do Not Call" List   mattf
• [Asterisk-Users] SNOM 200 and Asterisk Woes   Dan Mahoney, System Admin
• [Asterisk-Users] codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes from the end)   Yap Teong Eng
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   Martin List-Petersen
• [Asterisk-Users] Parked Call Viewer Application - Freeware   Brian McSpadden
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Mike Coakley
• [Asterisk-Users] Cisco 12sp+ and 30VIP   Mike Coakley
• [Asterisk-Users] Cisco 12sp+ and 30VIP   Carlos Hernandez
• [Asterisk-Users] The CISCO 7940 Tranfer Button..   James Gardiner
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   clive18 at webmail.co.za
• [Asterisk-Users] The CISCO 7940 Tranfer Button..   denon
• [Asterisk-Users] Cisco 12sp+ and 30VIP   Jeremy McNamara
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   Christoph Rothe
• [Asterisk-Users] SNOM 200 and Asterisk Woes   Olle E. Johansson
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   Peter Svensson
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   Soren Rathje
• [Asterisk-Users] Blocking the 'Do Not Call" List   Holger Schurig
• [Asterisk-Users] Blocking the 'Do Not Call" List   John Todd
• [Asterisk-Users] can sip users login two times?   Thomas Kuepper
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   John Todd
• [Asterisk-Users] RxFax - tiff file problem   Snezhana Bekova
• [Asterisk-Users] No cdr entries for calls coming via chan_oh323   Roger Schreiter
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   John Todd
• [Asterisk-Users] Blocking the 'Do Not Call" List   Steve Szmidt
• [Asterisk-Users] Zaptel Dial Out Issues   Bryan Vyhmeister
• [Asterisk-Users] RxFax - tiff file problem   irmantas.gudelis at vilnius2.balt.net
• [Asterisk-Users] can sip users login two times?   Thorsten Huber
• [Asterisk-Users] is gatekeeper required?   ml_asterisk-users at rajnoch.sk
• [Asterisk-Users] external mailbox   Stefan Märkle
• [Asterisk-Users] Blocking the 'Do Not Call" List   Senad Jordanovic
• [Asterisk-Users] Codec Problems   Nilesh sonavani
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   Soren Rathje
• [Asterisk-Users] 831 Santa Cruz/Watsoncille, Calif. DIDs   lists-jmhunter
• [Asterisk-Users] The CISCO 7940 Tranfer Button..   lists-jmhunter
• [Asterisk-Users] external mailbox   Eran Gal
• [Asterisk-Users] external mailbox   matt.riddell at sineapps.com
• [Asterisk-Users] Blocking the 'Do Not Call" List   Senad Jordanovic
• [Asterisk-Users] 2.4.x-SMP vs. 2.6.x-SMP   Bastian Schern
• [Asterisk-Users] stun and only one external ip   Thomas Kuepper
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   Peter Corlett
• [Asterisk-Users] Analog Phones with Status Light Indicators   Steven Critchfield
• [Asterisk-Users] SIP Transfers (Possibly reinvite)   Michael Graves
• [Asterisk-Users] Codec Problems   steve at daviesfam.org
• [Asterisk-Users] Blocking the 'Do Not Call" List   Peter Corlett
• [Asterisk-Users] Static on outgoing calls (Quad E1)   Ben Merrills
• [Asterisk-Users] Analog Phones with Status Light Indicators   Jeremy Lowery
• [Asterisk-Users] Ringing() doesn't play sound while phone is ringing   Warren Burstein
• [Asterisk-Users] zaphfc problems...   Gary Pigott
• [Asterisk-Users] zaphfc problems...   Andreas Sikkema
• [Asterisk-Users] Re: VoIP SPAM, what's next ?   Duane
• [Asterisk-Users] zaphfc problems...   Martin List-Petersen
• [Asterisk-Users] Static on outgoing calls (Quad E1)   Brent Franks
• [Asterisk-Users] The CISCO 7940 Tranfer Button..   Shaun Ewing
• [Asterisk-Users] zap channels and sip channels problem   Bartosz Jozwiak
• [Asterisk-Users] is gatekeeper required?   Roger Schreiter
• [Asterisk-Users] zaphfc problems...   Michael Sandee
• [Asterisk-Users] Cisco 12sp+ and 30VIP   Mike Coakley
• [Asterisk-Users] Cisco 12sp+ and 30VIP   Mike Coakley
• [Asterisk-Users] zaphfc problems...   Gary Pigott
• [Asterisk-Users] zaphfc problems...   Gary Pigott
• [Asterisk-Users] Waiting till picks up Zap line   Tom Lawrence
• [Asterisk-Users] Ringing() doesn't play sound while phone is ringing   Bart Coppens
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   Nate Carlson
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   Nate Carlson
• [Asterisk-Users] zaphfc problems...   Michael Welter
• [Asterisk-Users] The CISCO 7940 Tranfer Button..   Kubat, Philip
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Christopher L. Wade
• [Asterisk-Users] Polycom Echo   John Bittner
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   Bruce Komito
• [Asterisk-Users] 2.4.x-SMP vs. 2.6.x-SMP   Jean-Yves Avenard
• [Asterisk-Users] zaphfc problems...   Gary Pigott
• [Asterisk-Users] Static on outgoing calls (Quad E1)   Ben Merrills
• [Asterisk-Users] number unavailable   Thomas Kuepper
• [Asterisk-Users] asterisk -r and -rx questions   Andrew Kohlsmith
• [Asterisk-Users] Cisco 12sp+ and 30VIP   Mike Coakley
• [Asterisk-Users] Ringing() doesn't play sound while phone is ringing   Warren Burstein
• [Asterisk-Users] Static on outgoing calls (Quad E1)   Peter Svensson
• [Asterisk-Users] limit incoming calls to sip extens   Ryan Courtnage
• [Asterisk-Users] Snom Intercom   Ryan Ayers
• [Asterisk-Users] zaphfc problems...   Holger Schurig
• [Asterisk-Users] Asterisk and SMP   Matt Schulte
• [Asterisk-Users] zaphfc problems...   Michael Sandee
• [Asterisk-Users] Snom Intercom   Ariel's Hotmail
• [Asterisk-Users] zaphfc problems...   Gary Pigott
• [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?   Nate Carlson
• [Asterisk-Users] X100P outbound only (Don't answer)   David Cook
• [Asterisk-Users] 2.4.x-SMP vs. 2.6.x-SMP   Bastian Schern
• [Asterisk-Users] X100P outbound only (Don't answer)   Sean Cheesman
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Mike Coakley
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Christopher L. Wade
• [Asterisk-Users] Blocking the 'Do Not Call" List   Gabriel Millerd
• [Asterisk-Users] Static on outgoing calls (Quad E1)   Andres
• [Asterisk-Users] Blocking the 'Do Not Call" List   Peter Corlett
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Mike Coakley
• [Asterisk-Users] stun and only one external ip   Andres
• [Asterisk-Users] Polycom Echo   Patrick
• [Asterisk-Users] X100P outbound only (Don't answer)   Soren Rathje
• [Asterisk-Users] Blocking the 'Do Not Call" List   Chris Shaw
• [Asterisk-Users] zaphfc problems...   Scott Stingel
• [Asterisk-Users] Static on outgoing calls (Quad E1)   Ben Merrills
• [Asterisk-Users] Autoattendant Configuration   John Blackman
• [Asterisk-Users] Blocking the 'Do Not Call" List   Andrew Kohlsmith
• [Asterisk-Users] AbsoluteTimeout Inside A Macro   Christopher L. Wade
• [Asterisk-Users] a few question about asterisk   Brent Franks
• [Asterisk-Users] Polycom Echo   Tor Roberts
• [Asterisk-Users] Blocking the 'Do Not Call" List   Richard Lyman
• [Asterisk-Users] CallerID Debug On Zap/POTS Channel   Christopher L. Wade
• [Asterisk-Users] a few question about asterisk   Brassfield, Anthony
• [Asterisk-Users] Polycom Echo   John Baker
• [Asterisk-Users] Blocking the 'Do Not Call" List   Steven Critchfield
• [Asterisk-Users] a few question about asterisk   Steven Critchfield
• [Asterisk-Users] SNOM 200 and Asterisk Woes   Dan Mahoney, System Admin
• [Asterisk-Users] Fedora Core 2 (kernel 2.6.5), CAPI and Fritz PCI   John Daragon
• [Asterisk-Users] False Hangup detected on Digium TDM400P   Ruben Fagundo
• [Asterisk-Users] Static on outgoing calls (Quad E1)   Andres
• [Asterisk-Users] Comcast Phone Line hang up not detected   Ruben Fagundo
• [Asterisk-Users] CallerID Debug On Zap/POTS Channel   Soren Rathje
• [Asterisk-Users] CallerID Debug On Zap/POTS Channel   Christopher L. Wade
• [Asterisk-Users] Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted   miguel at amplanet.com.br
• [Asterisk-Users] Grandstream Budgetone-102 client cannot register   Michael Cheedle
• [Asterisk-Users] Re: X100P outbound only (Don't answer)   David Cook
• [Asterisk-Users] Avaya and Asterisk   Katerina Sadri
• [Asterisk-Users] Blind Call Transfer using Sipura 3000 + asterisk   Karun Chemudugunta
• [Asterisk-Users] Avaya IP Phones and Asterisk   Katerina Sadri
• [Asterisk-Users] Avaya IP Phones and Asterisk   Tenorio, Leandro
• [Asterisk-Users] Re: X100P outbound only (Don't answer)   Soren Rathje
• [Asterisk-Users] Problem with Audio Play out in Voice-mail   Tony Engel
• [Asterisk-Users] Inband announcement of parking slot from app_parkandannounce?   Kris Boutilier
• [Asterisk-Users] StanaPhone and Asterisks   Travis Conway
• [Asterisk-Users] Does IConnectHere still work with asterisk?   todd palumbo
• [Asterisk-Users] StanaPhone and Asterisks   Pulu 'Anau
• [Asterisk-Users] StanaPhone and Asterisks   Chris Shaw
• [Asterisk-Users] StanaPhone and Asterisks   Travis Conway
• [Asterisk-Users] Asterisk and SMP   Steve Szmidt
• [Asterisk-Users] Asterisk & Recommended Distro   Bryan Vyhmeister
• [Asterisk-Users] Asterisk & MyPhoneCompany.com (aka Talk(n))   Chris Shaw
• [Asterisk-Users] Grandstream Budgetone-102 client cannot register   Steve Szmidt
• [Asterisk-Users] Asterisk & MyPhoneCompany.com (aka Talk(n))   Travis Conway
• [Asterisk-Users] Does IConnectHere still work with asterisk?   Travis Conway
• [Asterisk-Users] H323 call dropped when answered   Krystian Filiks
• [Asterisk-Users] Asterisk & MyPhoneCompany.com (aka Talk(n))   Chris Shaw
• [Asterisk-Users] persistant SABME   aleph lammim
• [Asterisk-Users] Asterisk & Recommended Distro   Robert Barnes
• [Asterisk-Users] Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted (tethereal)   miguel at amplanet.com.br
• [Asterisk-Users] Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted (syslog)   miguel at amplanet.com.br
• [Asterisk-Users] persistant SABME   Rick L. Wilson, Sr.
• [Asterisk-Users] BroadVoice Voicemail   Chris
• [Asterisk-Users] Blocking the 'Do Not Call" List   Adam Goryachev
• [Asterisk-Users] Static on outgoing calls (Quad E1)   Adam Goryachev
• [Asterisk-Users] Starting with digium cards   Saurabh Garg
• [Asterisk-Users] BroadVoice Voicemail   Chris
• [Asterisk-Users] X-Lite behind NATed ADSL router   Araba, Michael
• [Asterisk-Users] BroadVoice Voicemail   Chris
• [Asterisk-Users] Analog Phones with Status Light Indicators   Adam Goryachev
• [Asterisk-Users] BroadVoice Voicemail   Jay Milk
• [Asterisk-Users] zaphfc problems...   Holger Schurig
• [Asterisk-Users] zaphfc capi problems...   Sascha Growe
• [Asterisk-Users] Static on outgoing calls (Quad E1)   Ben Merrills
• [Asterisk-Users] Static on outgoing calls (Quad E1)   Ben Merrills
• AW: [Asterisk-Users] Starting with digium cards   Karlheinz Hagen
• [Asterisk-Users] Starting with digium cards   Holger Schurig
• [Asterisk-Users] zaphfc problems...   Gary Pigott
• [Asterisk-Users] Pulse dialing...   Lev
• [Asterisk-Users] Decent Asterisk Compatible VoIP HardPhone - Australia   Stephen Allan
• [Asterisk-Users] Convert Cisco 7960 to sip   tmeeks at che-llp.com
• [Asterisk-Users] Pulse dialing...   Lev
• [Asterisk-Users] Convert Cisco 7960 to sip   Greg Smith
• [Asterisk-Users] zaphfc problems...   Scott Stingel
• [Asterisk-Users] Analog Phones with Status Light Indicators   Steven Critchfield
• [Asterisk-Users] zaphfc problems...   Gary Pigott
• [Asterisk-Users] Problems receiving SIP calls   Simon Brown
• [Asterisk-Users] Convert Cisco 7960 to sip   Simon Brown
• [Asterisk-Users] zaphfc problems...   Scott Stingel
• [Asterisk-Users] BRI and E1 in same system   Scott Stingel
• [Asterisk-Users] BRI and E1 in same system   Klaus-Peter Junghanns
• [Asterisk-Users] zaphfc problems...   Gary Pigott
• [Asterisk-Users] Analog Phones with Status Light Indicators   Adam Goryachev
• [Asterisk-Users] Convert Cisco 7960 to sip   asteriskstuff at ziplip.com
• [Asterisk-Users] BRI and E1 in same system   Scott Stingel
• [Asterisk-Users] Problems receiving SIP calls   Thorsten Huber
• [Asterisk-Users] Pulse dialing...   Lev
• [Asterisk-Users] Analog Phones with Status Light Indicators   Peter Svensson
• [Asterisk-Users] BroadVoice Voicemail   Greg Hill
• [Asterisk-Users] RES: Asterisk --> Mediatrix 1204 --> returned -1: Operation not permitted   miguel at amplanet.com.br
• [Asterisk-Users] DTMF issues   dbruce at bananatel.com
• [Asterisk-Users] Analog Phones with Status Light Indicators   Adam Goryachev
• [Asterisk-Users] Analog Phones with Status Light Indicators   Rich Adamson
• [Asterisk-Users] Problems receiving SIP calls   Thorsten Huber
• [Asterisk-Users] Autoattendant Configuration   Leif Madsen
• [Asterisk-Users] <<< MEETME_AGI_BACKGROUND inside MEET ME>>>   jeff quade
• [Asterisk-Users] Asterisk and SMP   Andrew Thompson
• [Asterisk-Users] Asterisk and SMP   Andrew Thompson
• [Asterisk-Users] Convert Cisco 7960 to sip   Joel Vandal
• [Asterisk-Users] H323 call dropped when answered   Krystian.Filiks
• [Asterisk-Users] Problems receiving SIP calls   Rich Adamson
• [Asterisk-Users] zaptel wont compile   AJ Grinnell
• [Asterisk-Users] Asterisk and SMP   Steve Szmidt
• [Asterisk-Users] Convert Cisco 7960 to sip   Dominique Kull
• [Asterisk-Users] zaptel wont compile   Travis Conway
• [Asterisk-Users] Blocking the 'Do Not Call" List   Steve Szmidt
• [Asterisk-Users] BroadVoice Voicemail   Greg Hill
• [Asterisk-Users] Asterisk & MyPhoneCompany.com (aka Talk(n))   Travis Conway
• [Asterisk-Users] Re: <<< MEETME_AGI_BACKGROUND inside MEET ME>>>   Tony Mountifield
• [Asterisk-Users] How Many Calls On This Config   Luke Catranis
• [Asterisk-Users] AgentLogin issue   Maurizio Marini
• [Asterisk-Users] Re: Convert Cisco 7960 to sip   Randy Bush
• [Asterisk-Users] caller id over iax   Stan Dard
• [Asterisk-Users] Asterisk & MyPhoneCompany.com (aka Talk(n))   Richard Cook
• [Asterisk-Users] RE: Autoattendant Configuration   John Blackman
• [Asterisk-Users] BroadVoice Voicemail   Chris Shaw
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #4901 - 10 msgs   Jason Kawakami
• [Asterisk-Users] re: Convert Cisco 7960 to sip   tmeeks at che-llp.com
• [Asterisk-Users] How Many Calls On This Config   mattf
• [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #4901 - 10 msgs   Chris Shaw
• [Asterisk-Users] Grandstream Budgetone-102 client cannot register   Michael Cheedle
• [Asterisk-Users] Asterisk & MyPhoneCompany.com (aka Talk(n))   Greg Hill
• [Asterisk-Users] TDM400P and Adtran 616 problems   Jason T. Nelson
• [Asterisk-Users] Blocking the 'Do Not Call" List   Dana Nowell
• [Asterisk-Users] voicetronix   Prof. Marcelo Kruk
• [Asterisk-Users] ::::: Pssst. Rc2! :::::::   Olle E. Johansson
• [Asterisk-Users] CCM <->(H323) <-> *   Andres Junge
• [Asterisk-Users] H323 call dropped when answered   administrator tootai
• [Asterisk-Users] New office hardware set up question.   Shawn Parker
• [Asterisk-Users] Problem installing Software Fax SpanDSP support into Asterisk   Jon Bebeau
• [Asterisk-Users] Blocking the 'Do Not Call" List   Richard Lyman
• [Asterisk-Users] zaptel wont compile   AJ Grinnell
• [Asterisk-Users] ::::: Pssst. Rc2! :::::::   Andrew Kohlsmith
• [Asterisk-Users] How Many Calls On This Config   Andrew Thompson
• [Asterisk-Users] <<< MEETME_AGI_BACKGROUND inside MEET ME>>>   jeff quade
• [Asterisk-Users] H323 call dropped when answered   Krystian Filiks
• [Asterisk-Users] Problem installing Software Fax SpanDSP support into Asterisk   Seth Remington
• [Asterisk-Users] Broadvoice User hung up on voicemail   Kevin
• [Asterisk-Users] Problems receiving SIP calls   Philip Fleischer
• [Asterisk-Users] Voicemail.conf not being re-read / Updates (e.g. pin numbers) ok.   Joshua McClintock
• [Asterisk-Users] Broadvoice User hung up on voicemail   Chris Shaw
• [Asterisk-Users] ::::: Pssst. Rc2! :::::::   Greg Boehnlein
• [Asterisk-Users] X100P a winmodem?   asteriskstuff at ziplip.com
• [Asterisk-Users] ::::: Pssst. Rc2! :::::::   Tim McKee
• [Asterisk-Users] Message lamp integration with legacy pbx -- revisited   Greg Blakely
• [Asterisk-Users] re: Convert Cisco 7960 to sip   asteriskstuff at ziplip.com
• [Asterisk-Users] Interruptable SayUnixTime   Dameon D. Welch-Abernathy
• [Asterisk-Users] Blocking the 'Do Not Call" List   Steven Critchfield
• [Asterisk-Users] Enterprise users of asterisk?   Brian Capouch
• [Asterisk-Users] Re: Convert Cisco 7960 to sip   asteriskstuff at ziplip.com
• [Asterisk-Users] Problem installing Software Fax SpanDSPsupport into Asterisk   Jon Bebeau
• [Asterisk-Users] ::::: Pssst. Rc2! :::::::   Andrew Kohlsmith
• [Asterisk-Users] How Many Calls On This Config   mattf
• [Asterisk-Users] Blind Call Transfer using Sipura 3000 + asterisk   Dameon D. Welch-Abernathy
• [Asterisk-Users] Re: <<< MEETME_AGI_BACKGROUND inside MEET ME>>>   Tony Mountifield
• [Asterisk-Users] X100P a winmodem?   Joaquin Cuenca Abela
• [Asterisk-Users] asterisk and rtp proxy   Ming-Wei Shih
• [Asterisk-Users] Default endian for signed linear   Jim O'Brien
• [Asterisk-Users] Zip2 configuration via tftp?   Bruce Komito
• [Asterisk-Users] H323 call dropped when answered   Krystian Filiks
• [Asterisk-Users] Broadvoice User hung up on voicemail   Kevin
• [Asterisk-Users] Problem installing Software Fax SpanDSPsupport into Asterisk   nospams
• [Asterisk-Users] GrandStream CallerID   nospams
• [Asterisk-Users] H323 call dropped when answered   administrator tootai
• [Asterisk-Users] Problem installing Software Fax SpanDSPsupport into Asterisk   Ed Brady
• [Asterisk-Users] Problem installing Software Fax SpanDSPsupport into Asterisk   Seth Remington
• [Asterisk-Users] ::::: Pssst. Rc2! :::::::   Steven Critchfield
• [Asterisk-Users] Asterisk and SER   Kurtz
• [Asterisk-Users] RE: Convert Cisco 7960 to sip   Greg Smith
• [Asterisk-Users] Question about TE405P   Angel Diaz
• [Asterisk-Users] Problems receiving SIP calls   Simon Brown
• [Asterisk-Users] caller id over iax   Stan Dard
• [Asterisk-Users] outgoing ZAP cannot connect using E1 isdn   Asterisk
• [Asterisk-Users] Calling Card App   Richard Cook
• [Asterisk-Users] H323 problems   Krystian Filiks
• [Asterisk-Users] Blind Call Transfer using Sipura 3000 + aste risk   Karun Chemudugunta
• [Asterisk-Users] First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs?   Brian Elton
• [Asterisk-Users] Avaya IP Phones and Asterisk   Brian Elton
• [Asterisk-Users] Avaya and Asterisk   Brian Elton
• [Asterisk-Users] AgentLogin issue   Joe Dennick
• [Asterisk-Users] Blocking the 'Do Not Call" List   Adam Goryachev
• [Asterisk-Users] Grandstream HandyTone-486 FXO port   Stefaan
• [Asterisk-Users] Problems receiving SIP calls   Simon Brown
• [Asterisk-Users] RE: Convert Cisco 7960 to sip   Bryan Vyhmeister
• [Asterisk-Users] RE: Convert Cisco 7960 to sip   Brian McManus
• [Asterisk-Users] H323 call dropped when answered   Krystian.Filiks
• [Asterisk-Users] Grandstream Budgetone-102 client cannot register   Holger Schurig
• [Asterisk-Users] outgoing ZAP cannot connect using E1 isdn   Peter Svensson
• [Asterisk-Users] Interruptable SayUnixTime   Holger Schurig
• [Asterisk-Users] outgoing ZAP cannot connect using E1 isdn   Asterisk
• [Asterisk-Users] outgoing ZAP cannot connect using E1 isdn   Asterisk
• [Asterisk-Users] Asterisk and SER   Asterisk .
• [Asterisk-Users] outgoing ZAP cannot connect using E1 isdn   Peter Svensson
• [Asterisk-Users] incomingcall braking all   Altus Snyman
• [Asterisk-Users] RC2 - H323 channel broken   administrator tootai
• [Asterisk-Users] RC2 - H323 channel broken   T. Chan
• [Asterisk-Users] Question about TE405P   Claus Futtrup
• [Asterisk-Users] AgentLogin issue   Maurizio Marini
• [Asterisk-Users] ZAPHFC for Euro-ISDN   Bastian Schern
• [Asterisk-Users] * -> Lucent PBX   Dave Cotton
• [Asterisk-Users] H323 call dropped when answered   Asterisk .
• [Asterisk-Users] Problem with grandstream devices and DTMF signalling   Simone Ricci
• [Asterisk-Users] Re: Send DTMF tone Like 'C' on connected call   Antonio Rabena
• [Asterisk-Users] Question about TE405P   Scott Stingel
• [Asterisk-Users] Re: Send DTMF tone Like 'C' on connected call   Simone Ricci
• [Asterisk-Users] Question about TE405P   Scott Stingel
• [Asterisk-Users] not hangup   Altus Snyman
• [Asterisk-Users] H323 call dropped when answered   Krystian.Filiks
• [Asterisk-Users] queue name too long when sending sms over 32 chars   daschi at gmx.net
• [Asterisk-Users] Lost 7960 time display on upgrade   David Cook
• [Asterisk-Users] H323 call dropped when answered   Krystian.Filiks
• [Asterisk-Users] Problem with grandstream devices and DTMF signalling [RESOLVED]   Simone Ricci
• [Asterisk-Users] Need help for GUI Clieny   Dhaval mehta
• [Asterisk-Users] Need help for GUI Client   Dhaval mehta
• [Asterisk-Users] Problem with grandstream devices and DTMF signalling   Holger Schurig
• [Asterisk-Users] not hangup   Christoph Rothe
• [Asterisk-Users] asterisk in india   Vikram Rangnekar
• [Asterisk-Users] not hangup   Altus Snyman
• [Asterisk-Users] OH.323 Dialout Problem   Brian Wilkins
• [Asterisk-Users] Asterisk and SER   Miroslav Nachev
• [Asterisk-Users] ::::: Pssst. Rc2! :::::::   Andrew Kohlsmith
• [Asterisk-Users] Interop RTP "Extension headers" for QOS?   Matt Schulte
• [Asterisk-Users] te410p and Telstra Onramp 10   Craig Guy
• [Asterisk-Users] X100P a winmodem?   Rich Adamson
• [Asterisk-Users] Re: Grandstream Budgetone-102 client cannot register   Stephen R. Besch
• [Asterisk-Users] small asterisk system   albayrak at lycos.co.uk
• [Asterisk-Users] Lost 7960 time display on upgrade   Leif Madsen
• [Asterisk-Users] small asterisk system   Lyle Giese
• [Asterisk-Users] SpanDSP - Training failed (convergence failed) error   Stephen Allan
• [Asterisk-Users] SIP <->h.323   Yiannis Costopoulos, Web2Net Solutions Ltd.
• [Asterisk-Users] Static on outgoing calls (Quad E1)   Ben Merrills
• [Asterisk-Users] MGCP & RFC3149 & Swissvoice IP10   Bennett, Chris
• [Asterisk-Users] SpanDSP - Training failed (convergence failed) error   Holger Schurig
• [Asterisk-Users] Interop RTP "Extension headers" for QOS?   Matt Schulte
• [Asterisk-Users] Analog Phones with Status Light Indicators   Rich Adamson
• [Asterisk-Users] H323 call dropped when answered   Asterisk .
• [Asterisk-Users] SIP <->h.323   Ryan Wilkins
• [Asterisk-Users] asterisk in india   Asterisk .
• [Asterisk-Users] Asterisk and SER   Asterisk .
• [Asterisk-Users] asterisk & gnugk & voip phone   Maros RAJNOCH
• [Asterisk-Users] Broadvoice User hung up on voicemail   Chris Shaw
• [Asterisk-Users] Lost 7960 time display on upgrade   Rich Adamson
• [Asterisk-Users] Playback doesn't work whith h323   M. Willigs
• [Asterisk-Users] voicemail email messages   defiance
• [Asterisk-Users] Will this ISDN card work for me?   Rob Fugina
• [Asterisk-Users] Broadvoice User hung up on voicemail   Chris Shaw
• [Asterisk-Users] Re: Lost 7960 time display on upgrade   David Cook
• [Asterisk-Users] Problem with ougoing Zap calls   Russell Horn
• [Asterisk-Users] Problem with ougoing Zap calls   Asterisk
• [Asterisk-Users] [Asterisk-Users]SIP<->H323 "Failed to create smoother"   Zineddin Karzazi
• [Asterisk-Users] Broadvoice User hung up on voicemail   Chris Shaw
• [Asterisk-Users] Broadvoice User hung up on voicemail   Kevin
• [Asterisk-Users] Cisco 79xx series IP phones   Shawn Parker
• [Asterisk-Users] Broadvoice User hung up on voicemail   Kevin
• [Asterisk-Users] *** Asterisk Summer News: Forget numbers, dial by domain!   Olle E. Johansson
• [Asterisk-Users] Broadvoice User hung up on voicemail   Chris Shaw
• [Asterisk-Users] Broadvoice User hung up on voicemail   Chris Shaw
• [Asterisk-Users] Cisco 79xx series IP phones   Brian D'Arcy
• [Asterisk-Users] Cisco 79xx series IP phones   Joshua M. Thompson
• [Asterisk-Users] How to detect answering machine   Christian Victor
• [Asterisk-Users] External MW Lamp On/Off   Greg Blakely
• [Asterisk-Users] Will this ISDN card work for me?   Patrick
• [Asterisk-Users] Will this ISDN card work for me?   Martin List-Petersen
• [Asterisk-Users] Broadvoice User hung up on voicemail   James Jones
• [Asterisk-Users] Cisco 79xx series IP phones   steveb at asbc.demon.co.uk
• [Asterisk-Users] Dial command problems   Michael Little
• [Asterisk-Users] Will this ISDN card work for me?   Lyle Giese
• [Asterisk-Users] Cisco 79xx series IP phones   Scott Laird
• [Asterisk-Users] Cisco 79xx series IP phones   Shawn Parker
• [Asterisk-Users] Cisco 79xx series IP phones   Gary Carr
• [Asterisk-Users] How to detect answering machine   Jay Milk
• [Asterisk-Users] How to detect answering machine   Brian Jones
• [Asterisk-Users] Euro isdn caller id problem   Asterisk
• [Asterisk-Users] Avaya and Asterisk   Tenorio, Leandro
• [Asterisk-Users] Euro isdn caller id problem   Peter Svensson
• [Asterisk-Users] Dial command problems   Lyle Giese
• [Asterisk-Users] voice choppy   Dana Nowell
• [Asterisk-Users] Dial command problems   nospams
• [Asterisk-Users] Euro isdn caller id problem   Asterisk
• [Asterisk-Users] Cisco 79xx series IP phones   asteriskstuff at ziplip.com
• [Asterisk-Users] Cisco 79xx series IP phones   Shawn Parker
• [Asterisk-Users] How to detect answering machine   Walt Reed
• [Asterisk-Users] Euro isdn caller id problem   Peter Svensson
• [Asterisk-Users] queue name too long when sending sms over 32 chars   Stefan Reuter
• [Asterisk-Users] Cisco 79xx series IP phones   Rich Adamson
• [Asterisk-Users] Asterisk and softswitch   Jorge Verastegui
• [Asterisk-Users] voice choppy   Luke Catranis
• [Asterisk-Users] voice choppy   Mike Benoit
• [Asterisk-Users] OH.323 Dialout Problem   administrator tootai
• [Asterisk-Users] Euro isdn caller id problem   Asterisk
• [Asterisk-Users] Dial command problems   Michael Little
• [Asterisk-Users] Dial command problems   Michael Little
• [Asterisk-Users] asterisk in india   Kanuri, Seshu
• [Asterisk-Users] voice choppy   Dana Nowell
• [Asterisk-Users] Static on outgoing calls using either X100P or TDM400P   Lenny Self
• [Asterisk-Users] asterisk in india   Kannaiyan Natesan
• [Asterisk-Users] SIP in mysql and reinvite   Ming-Wei Shih
• [Asterisk-Users] Blocking the 'Do Not Call" List   Peter Corlett
• [Asterisk-Users] Cisco Router as Gateway for FXO Ports   Bryan Vyhmeister
• [Asterisk-Users] voice choppy   Patrick
• [Asterisk-Users] Polycom IP600 shared lines   Chris HARIGA
• [Asterisk-Users] Using a TE405P to connect to an existing PBX   drodden
• [Asterisk-Users] Dial command problems   Lyle Giese
• [Asterisk-Users] voice choppy   Dana Nowell
• [Asterisk-Users] Euro isdn caller id problem   Peter Svensson
• [Asterisk-Users] Vlan question   tmeeks at che-llp.com
• [Asterisk-Users] Re: Dial command problems   Jason Kawakami
• [Asterisk-Users] voice choppy   Chris Shaw
• [Asterisk-Users] Using a TE405P to connect to an existing PBX   Peter Svensson
• [Asterisk-Users] Dial command problems   Trevor Peirce
• [Asterisk-Users] Vlan question   Bryan Vyhmeister
• [Asterisk-Users] Vlan question   Steven P. Donegan
• [Asterisk-Users] te410p and Telstra Onramp 10   David
• [Asterisk-Users] [Q] DIDs   Roderick A. Anderson
• [Asterisk-Users] [Q] DIDs   Chris Shaw
• [Asterisk-Users] Using a TE405P to connect to an existing PBX   drodden
• [Asterisk-Users] HELP: BYE-request not sent to SIP-peer   Roland Zagler
• [Asterisk-Users] Channel Bank   Daniel Bichara
• [Asterisk-Users] Channel Bank   Mitchel Constantin
• [Asterisk-Users] limit incoming calls to sip extens   Erik Barker
• [Asterisk-Users] te410p and Telstra Onramp 10   Adam Goryachev
• [Asterisk-Users] te410p and Telstra Onramp 10   Craig Guy
• [Asterisk-Users] te410p and Telstra Onramp 10   Craig Guy
• [Asterisk-Users] [Q] DIDs   Peter Svensson
• [Asterisk-Users] Using a TE405P to connect to an existing PBX   Peter Svensson
• [Asterisk-Users] Vlan question   Florian Overkamp
• [Asterisk-Users] SIP <->h.323   Bernie Hoeneisen
• [Asterisk-Users] [Q] DIDs   Robert Hajime Lanning
• [Asterisk-Users] [Q] DIDs   Peter Svensson
• [Asterisk-Users] te410p and Telstra Onramp 10   Craig Guy
• [Asterisk-Users] External MW Lamp On/Off   Umar Sear
• [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?   Andy Lee
• [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?   Lubomir Christov
• [Asterisk-Users] Howto remove digits from a called number   administrator tootai
• [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?   Dennis Cartier
• [Asterisk-Users] Howto remove digits from a called number   Greg Hill
• [Asterisk-Users] Confused --> Hardware specs   Vasyl Rublyov
• [Asterisk-Users] List traffic/Software   ast at uc9.net
• [Asterisk-Users] External MW Lamp On/Off   Greg Blakely
• [Asterisk-Users] Questions   Ildar Gabdulline
• [Asterisk-Users] Re: Questions   Tony Mountifield
• [Asterisk-Users] Help - is voip good for in-house calls?   Francis Augusto Medeiros
• [Asterisk-Users] Help - is voip good for in-house calls?   Greg Broiles
• [Asterisk-Users] Help - is voip good for in-house calls?   Wiley E. Siler
• [Asterisk-Users] BugetTone Bug Showstopper,   michael koehler
• [Asterisk-Users] Polycom IP600 shared lines   John Baker
• [Asterisk-Users] Help - is voip good for in-house calls?   Francis Augusto Medeiros
• [Asterisk-Users] Help - is voip good for in-house calls?   Francis Augusto Medeiros
• [Asterisk-Users] Installing Zaptel Modules on Fedora Core 2   Vikas Deolaliker
• [Asterisk-Users] Help - is voip good for in-house calls?   Peter Svensson
• [Asterisk-Users] Help - is voip good for in-house calls?   Wiley E. Siler
• [Asterisk-Users] Help - is voip good for in-house calls?   Peter Svensson
• [Asterisk-Users] Help - is voip good for in-house calls?   Peter Svensson
• [Asterisk-Users] Help - is voip good for in-house calls?   Francis Augusto Medeiros
• [Asterisk-Users] Help - is voip good for in-house calls?   Francis Augusto Medeiros
• [Asterisk-Users] Help - is voip good for in-house calls?   Andrew Kohlsmith
• [Asterisk-Users] Free MOH MP3   Wiley E. Siler
• [Asterisk-Users] Free MOH MP3   asterisk at bsius.com
• [Asterisk-Users] Free MOH MP3   Wiley E. Siler
• [Asterisk-Users] Polycom IP600 shared lines   Chris HARIGA
• [Asterisk-Users] Questions on various and sundry IP phones, and cabling   Adolf Osborne
• [Asterisk-Users] List traffic/Software   Matt Schulte
• [Asterisk-Users] Analog Phones with Status Light Indicators   Adam Goryachev
• [Asterisk-Users] Polycom IP600 shared lines   John Baker
• [Asterisk-Users] Pthread issue   Johnathan Bunn
• [Asterisk-Users] 7960 help   Jason Kawakami
• [Asterisk-Users] 7960 help   Gonzalo Gasca Meza
• [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?   MPlus
• [Asterisk-Users] Help - is voip good for in-house calls?   Peter Svensson
• [Asterisk-Users] Help - is voip good for in-house calls?   Peter Svensson
• [Asterisk-Users] Help - is voip good for in-house calls?   Jay Milk
• [Asterisk-Users] List traffic/Software   Jay Milk
• [Asterisk-Users] Using a TE405P to connect to an existing PBX   Deon Rodden
• [Asterisk-Users] Inbound Free World Dialup - extension not ringing?   Chris Blunt
• [Asterisk-Users] Do you speak Czech?   Olle E. Johansson
• [Asterisk-Users] Help - is voip good for in-house calls?   Tobias Jönsson
• [Asterisk-Users] chan_oh323 loading error   Krystian Filiks
• [Asterisk-Users] Vlan question   asteriskstuff at ziplip.com
• [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?   Dennis Cartier
• [Asterisk-Users] Help - is voip good for in-house calls?   Francis Augusto Medeiros
• [Asterisk-Users] Help - is voip good for in-house calls?   Peter Svensson
• [Asterisk-Users] Sip to Sip Calls via Asterisk   David Allen
• [Asterisk-Users] Installing Zaptel Modules on Fedora Core 2   Leif Madsen
• [Asterisk-Users] Installing Zaptel Modules on Fedora Core 2   Craig Guy
• [Asterisk-Users] Inbound Free World Dialup - extension not ringing?   Lyle Giese
• [Asterisk-Users] chan_oh323 loading error   Ryan Wilkins
• [Asterisk-Users] chan_oh323 loading error   Krystian Filiks
• [Asterisk-Users] chan_oh323 loading error   Krystian Filiks
• [Asterisk-Users] CCM <->(H323) <-> *   Chris Luke
• [Asterisk-Users] Howto remove digits from a called number   administrator tootai
• [Asterisk-Users] chan_oh323 loading error   Ryan Wilkins
• [Asterisk-Users] Free MOH MP3   William Suffill
• [Asterisk-Users] Help - is voip good for in-house calls?   Nicolas Gudino
• [Asterisk-Users] Vlan question   Steve Szmidt
• [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?   MPlus
• [Asterisk-Users] Free MOH MP3   Peter Corlett
• [Asterisk-Users] chan_oh323 loading error   Krystian Filiks
• [Asterisk-Users] Help - is voip good for in-house calls?   Francis Augusto Medeiros
• [Asterisk-Users] chan_oh323 loading error   Ryan Wilkins
• [Asterisk-Users] chan_oh323 loading error   Krystian Filiks
• [Asterisk-Users] chan_oh323 loading error   Ryan Wilkins
• [Asterisk-Users] Help - is voip good for in-house calls?   Andrew Kohlsmith
• [Asterisk-Users] New $89 VOIP phone   James H Thompson
• [Asterisk-Users] SIP connections do not hang up   Ian Hailey
• [Asterisk-Users] Help - is voip good for in-house calls?   Francis Augusto Medeiros
• [Asterisk-Users] GrandStream ATA286 & RC2 (was RC2 - H323 channel broken)   administrator tootai
• [Asterisk-Users] Asterisk MIBS   Alagalah
• [Asterisk-Users] Free MOH MP3   Kevin Walsh
• [Asterisk-Users] Help - is voip good for in-house calls?   Andrew Kohlsmith
• [Asterisk-Users] chan_oh323 loading error   Krystian Filiks
• [Asterisk-Users] Free MOH MP3   William Suffill
• [Asterisk-Users] asterisk with InPhonex?   hank
• [Asterisk-Users] Free MOH MP3   Kevin Walsh
• [Asterisk-Users] Re: Free MOH MP3   Tony Mountifield
• [Asterisk-Users] Re:Re:7960 help   Jason Kawakami
• [Asterisk-Users] Internal Distinctive Ringing + Caller ID   Greg Blakely
• [Asterisk-Users] __use_ast_pthread_create_instead__   Krystian Filiks
• [Asterisk-Users] Asterisk MIBS   Olle E. Johansson
• [Asterisk-Users] GrandStream ATA286 & RC2 (was RC2 - H323 channel broken)   administrator tootai
• [Asterisk-Users] Asterisk MIBS   Soren Rathje
• [Asterisk-Users] no tones detected   Johnathan Bunn
• [Asterisk-Users] asterisk+radius   mohammad
• [Asterisk-Users] no tones detected   Greg Hill
• [Asterisk-Users] Inbound Free World Dialup - extension not ringing?   Chris Blunt
• [Asterisk-Users] no tones detected   Johnathan Bunn
• [Asterisk-Users] Teliax TOS copied from Vonage?   Brad Ediger
• [Asterisk-Users] Inbound Free World Dialup - extension not ringing?   Lyle Giese
• [Asterisk-Users] Inbound Free World Dialup - extension not ringing?   Edward Eastman
• [Asterisk-Users] Re: GS HandyTone-286 Transfer Problem, can anyone confirm?   Andy Lee
• [Asterisk-Users] Newbie with missing .conf files   Don Moskaluk
• [Asterisk-Users] 123 Basic configuration files   Don Moskaluk
• [Asterisk-Users] 123 Basic configuration files   Wiley E. Siler
• [Asterisk-Users] Newbie with missing .conf files   Brian
• [Asterisk-Users] Help - is voip good for in-house calls?   Francis Augusto Medeiros
• [Asterisk-Users] Asterisk MIBS   Michael Welter
• [Asterisk-Users] Help - is voip good for in-house calls?   Nicolas Gudino
• [Asterisk-Users] how can i config a Cisco IAD 2430 config as a sip client   Jannette Mejia
• [Asterisk-Users] 123 Basic configuration files   Dan Clark
• [Asterisk-Users] Modified Prepaid doesn't update the balance   Glynn Condez
• [Asterisk-Users] consultative transfer with zaptel   Lars Lech
• [Asterisk-Users] Differences in CDR files   Mickey Binder
• [Asterisk-Users] Help: SMS in UK   Scott Stingel
• [Asterisk-Users] opencall.org down?   ePyron Felix Deierlein
• [Asterisk-Users] Static on outgoing calls using either X100P or TDM400P   Ben Merrills
• [Asterisk-Users] ## transfer into CVS when? Plus Suggestion. Attendant Transfer possible..   James Gardiner
• [Asterisk-Users] consultative transfer with zaptel   Holger Schurig
• [Asterisk-Users] Re: consultative transfer with zaptel   Tony Mountifield
• [Asterisk-Users] OH.323 Dialout Problem   Brian Wilkins
• [Asterisk-Users] Is "Meetme" a generic term?   Tony Mountifield
• [Asterisk-Users] Is "Meetme" a generic term?   Scott Stingel
• [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *   Olle E. Johansson
• [Asterisk-Users] Help - is voip good for in-house calls?   Holger Schurig
• [Asterisk-Users] New $89 VOIP phone   Steve Szmidt
• [Asterisk-Users] Help - is voip good for in-house calls?   Holger Schurig
• [Asterisk-Users] 123 Basic configuration files   Holger Schurig
• [Asterisk-Users] New $89 VOIP phone   Holger Schurig
• [Asterisk-Users] Free MOH MP3   Holger Schurig
• [Asterisk-Users] Using Asterisk in home environment - room with speakers&mic as IP phone ?   Robert Rozman
• [Asterisk-Users] Call stealing   Ben Merrills
• [Asterisk-Users] Re: consultative transfer with zaptel   Martin List-Petersen
• [Asterisk-Users] disable console channels   Michael George
• [Asterisk-Users] disable console channels   Andrew Kohlsmith
• [Asterisk-Users] disable console channels   Tobias Jönsson
• [Asterisk-Users] world incoming calls   Sergey Lapin
• [Asterisk-Users] Free MOH MP3   Matt Schulte
• [Asterisk-Users] no hangup   Altus Snyman
• [Asterisk-Users] Using Asterisk in home environment - room with speakers&mic as IP phone ?   Holger Schurig
• [Asterisk-Users] DTMoE crash the Asterisk and the Network   Miroslav Nachev
• [Asterisk-Users] world incoming calls   Olle E. Johansson
• [Asterisk-Users] Inbound Free World Dialup - extension not ringing?   Chris Blunt
• [Asterisk-Users] randomize Dial() target   Michael Devenijn
• [Asterisk-Users] randomize Dial() target   Marcin Mazurek
• [Asterisk-Users] What is the best Linux for asterisk   Johannes van Hulst
• [Asterisk-Users] randomize Dial() target   Edward Eastman
• [Asterisk-Users] What is the best Linux for asterisk   Matt Schulte
• [Asterisk-Users] Call stealing   Nicolas Gudino
• [Asterisk-Users] What is the best Linux for asterisk   Lyle Giese
• [Asterisk-Users] Using a TE405P to connect to an existing PBX   drodden
• [Asterisk-Users] local echo using SPA-3000 as FXO port   Michael Graves
• [Asterisk-Users] Using a TE405P to connect to an existing PBX   Andrew Kohlsmith
• [Asterisk-Users] randomize Dial() target   Marcin Mazurek
• [Asterisk-Users] dialing out and ringing issue   Info
• [Asterisk-Users] OH.323 Dialout Problem   administrator tootai
• [Asterisk-Users] What is the best Linux for asterisk   Vlok Stone
• [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?   James Freire
• [Asterisk-Users] Re: dialing out and ringing issue   Stephen R. Besch
• [Asterisk-Users] New $89 VOIP phone   Vlok Stone
• [Asterisk-Users] dialing out and ringing issue   Greg Hill
• [Asterisk-Users] Re: GrandStream ATA286 & RC2 (was RC2 - H323 channel broken)   administrator tootai
• [Asterisk-Users] Static on outgoing calls using either X100P or TDM400P   Ryan Courtnage
• [Asterisk-Users] ## transfer into CVS when? Plus Suggestion. Attendant Transfer possible..   hank
• [Asterisk-Users] New $89 VOIP phone   Richard Cook
• [Asterisk-Users] New $89 VOIP phone   Simone Ricci
• [Asterisk-Users] New $89 VOIP phone   Steven Critchfield
• [Asterisk-Users] __use_ast_pthread_create_instead__   Krystian.Filiks
• [Asterisk-Users] Static on outgoing calls using either X100P or TDM400P   Ryan Courtnage
• [Asterisk-Users] Compile error on Zaptel with Suse 9.1 (follow-up of subject: What is the best Linux for asterisk)   Johannes van Hulst
• [Asterisk-Users] New $89 VOIP phone   Chris Shaw
• [Asterisk-Users] New $89 VOIP phone   Steve Szmidt
• [Asterisk-Users] Using a TE405P to connect to an existing PBX   drodden
• [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?   Olle E. Johansson
• [Asterisk-Users] Static on outgoing calls using either X100P or TDM400P   Ben Merrills
• [Asterisk-Users] Faxdetect does not detect certain Fax machines   Frank
• [Asterisk-Users] Using a TE405P to connect to an existing PBX   Andrew Kohlsmith
• [Asterisk-Users] What is the best Linux for asterisk   Johnathan Bunn
• [Asterisk-Users] Using a TE405P to connect to an existing PBX   drodden
• [Asterisk-Users] 1.0 RC2 External AGI Issues   William R. Lorenz
• [Asterisk-Users] Re: 1.0 RC2 External AGI Issues   William R. Lorenz
• [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?   James Freire
• [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?   Greg Hill
• [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?   Sean Cheesman
• [Asterisk-Users] What is the best Linux for asterisk   Shawn Parker
• [Asterisk-Users] Free MOH MP3   hank
• [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?   Olle E. Johansson
• [Asterisk-Users] Re: Free MOH MP3   Leif Madsen
• [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?   Greg Hill
• [Asterisk-Users] Avaya firmware   Aaron Johnson
• [Asterisk-Users] Problems compiling chan_capi-0.3.5   Markus Engelbrecht
• [Asterisk-Users] DID Questions   Mike Roberts
• [Asterisk-Users] DID Questions   Mohammed Salim
• [Asterisk-Users] DID Questions   Mike Roberts
• [Asterisk-Users] DID Questions   Eric Wieling
• [Asterisk-Users] Set-up for 2 Asterisk System   Craig Neumanns
• [Asterisk-Users] DID Questions   Mike Roberts
• [Asterisk-Users] Set-up for 2 Asterisk System   Brian McManus
• [Asterisk-Users] DID Questions   Eric Wieling
• [Asterisk-Users] DID Questions   Mike Roberts
• [Asterisk-Users] DID Questions   Brian McManus
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   Derek Listmail Acct
• [Asterisk-Users] Problem compiling chan_sccp   Lex Lethol
• [Asterisk-Users] DID Questions   Greg Hill
• [Asterisk-Users] DID Questions   Tenorio, Leandro
• [Asterisk-Users] DID Questions   Mohammed Salim
• [Asterisk-Users] voicemail email messages   usedcanon
• [Asterisk-Users] Performance testing of asterisk   Tom Masterson
• [Asterisk-Users] External MW Lamp On/Off   usedcanon
• [Asterisk-Users] asterisk in india   usedcanon
• [Asterisk-Users] 1.0 RC2 External AGI Issues   usedcanon
• [Asterisk-Users] Call stealing   William Suffill
• [Asterisk-Users] voicemail email messages   defiance
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   Wiley E. Siler
• [Asterisk-Users] DID Questions   Mike Roberts
• [Asterisk-Users] DID Questions   Mike Roberts
• [Asterisk-Users] disable console channels   Michael George
• [Asterisk-Users] * and answering machine   Michael George
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   Patrick
• [Asterisk-Users] Soft DSS for Asterisk   Wiley E. Siler
• [Asterisk-Users] voicemail email messages   usedcanon
• [Asterisk-Users] Problem compiling chan_sccp   Lex Lethol
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   Wiley E. Siler
• [Asterisk-Users] Problems compiling chan_capi-0.3.5   Patrick
• [Asterisk-Users] Compile error on Zaptel with Suse 9.1 (follow-up of subject: What is the best Linux for asterisk)   Trevor G. Hammonds
• [Asterisk-Users] Music on hold question..   Ferrara, Jamie
• [Asterisk-Users] Performance testing of asterisk   Chris A. Icide
• [Asterisk-Users] How to run qozap with wcfxo and wsfxs together?   Maciej Kietlinski
• [Asterisk-Users] mysql version of Directory app   Bruce Komito
• [Asterisk-Users] * and answering machine   Rich Adamson
• [Asterisk-Users] RC2: Where is parking.conf and MeetMe app ?   Robert Rozman
• [Asterisk-Users] Performance testing of asterisk   Tom Masterson
• [Asterisk-Users] Digium TE410P and RedHat Enterprise Server 3.0   Roland Zagler
• [Asterisk-Users] SpanDSP - Training failed error / timing problem   Stephen Allan
• [Asterisk-Users] HELP with HFC Card, please   Sascha Growe
• [Asterisk-Users] * and answering machine   Michael George
• [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?   John Todd
• [Asterisk-Users] taking asterisk out of nat?   hank
• [Asterisk-Users] Teliax TOS copied from Vonage?   John Todd
• [Asterisk-Users] Answer a specific channel   Ravi Hanumaiah
• [Asterisk-Users] taking asterisk out of nat?   Steve Totaro
• [Asterisk-Users] asterisk in india   Steve Totaro
• [Asterisk-Users] taking asterisk out of nat?   hank
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   John Todd
• [Asterisk-Users] asterisk in india   Martin List-Petersen
• [Asterisk-Users] taking asterisk out of nat?   Andrew Kohlsmith
• [Asterisk-Users] DID Questions   Mohammed Salim
• [Asterisk-Users] How to use the Return Code???   James Gardiner
• [Asterisk-Users] Re: using mysql and asterisk   Dhaval mehta
• [Asterisk-Users] DID Questions   Greg Hill
• [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?   Greg Hill
• [Asterisk-Users] New $89 VOIP phone   Jay Milk
• [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?   Chris A. Icide
• [Asterisk-Users] RC2 - RH9   Martin Keding
• [Asterisk-Users] Formatting in sip.conf...can you have 2 @ signs for register?   Olle E. Johansson
• [Asterisk-Users] consultative transfer with zaptel   Lars Lech
• [Asterisk-Users] Digium TE410P and RedHat Enterprise Server 3.0   Peter Svensson
• [Asterisk-Users] (no subject)   yair at hakak.com
• [Asterisk-Users] asterisk in india   Navnit Chachan
• [Asterisk-Users] re: asterisk as VM for SER   yair at hakak.com
• [Asterisk-Users] RC2: Where is parking.conf and MeetMe app ?   Holger Schurig
• [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released   Shaun Ewing
• [Asterisk-Users] Problem compiling chan_sccp   Julien Goodwin
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   Michael Devenijn
• [Asterisk-Users] New $89 VOIP phone   Simone Ricci
• [Asterisk-Users] New $89 VOIP phone   Peter Svensson
• [Asterisk-Users] dropouts   Roy Sigurd Karlsbakk
• [Asterisk-Users] TDMoE crash the Asterisk and the Network   Miroslav Nachev
• [Asterisk-Users] Problem compiling chan_sccp   Lex Lethol
• [Asterisk-Users] New $89 VOIP phone   Simone Ricci
• [Asterisk-Users] Problems with DTMF   Simone Ricci
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   Patrick
• [Asterisk-Users] Performance testing of asterisk   Patrick
• [Asterisk-Users] Digium TE410P and RedHat Enterprise Server 3.0   Murali
• [Asterisk-Users] Problem compiling chan_sccp   Lex Lethol
• [Asterisk-Users] Performance testing of asterisk   Scott Stingel
• [Asterisk-Users] spandsp + files   Manfred Petz
• [Asterisk-Users] RC2: Where is parking.conf and MeetMe app ?   Robert Rozman
• [Asterisk-Users] has no CRC! error messages while compiling zaptel   Lars Lech
• [Asterisk-Users] DID Questions   Christian Victor
• [Asterisk-Users] New $89 VOIP phone   Steve Totaro
• [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released   Michael Løjtnant
• [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 release d   Low, Adam
• [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 release d   Low, Adam
• [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released - solved   Michael Løjtnant
• [Asterisk-Users] (no subject)   Mayank Mishra
• [Asterisk-Users] RE: dialing out   Info
• [Asterisk-Users] Compiling Zaptel under Bering 1.2   dmcintosh
• [Asterisk-Users] Faxing over ulaw   Matt Schulte
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   John Baker
• [Asterisk-Users] Faxing over ulaw   Andrew Kohlsmith
• [Asterisk-Users] Inbound IAX2 calls has no music on hold   Steve Szmidt
• [Asterisk-Users] RE: RE: dialing out   Info
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   Eric Wieling
• [Asterisk-Users] asterisk-wide variables   Kramer, R.D.J.
• [Asterisk-Users] CPC on Zaptel   Olle E. Johansson
• [Asterisk-Users] CVS Number after Update   Martin Keding
• [Asterisk-Users] Avaya firmware   Aaron Johnson
• [Asterisk-Users] CVS Number after Update   Olle E. Johansson
• [Asterisk-Users] CVS Number after Update   administrator tootai
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   Derek Listmail Acct
• [Asterisk-Users] BroadVOX   Dan Mahoney, System Admin
• [Asterisk-Users] Compile error on Zaptel with Suse 9.1 (follow-up of subject: What is the best Linux for asterisk)   Johannes van Hulst
• [Asterisk-Users] Re: asterisk in india   Vikram Rangnekar
• [Asterisk-Users] New $89 VOIP phone   Kanuri, Seshu
• [Asterisk-Users] OH323 and G729   Chris A. Icide
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   Derek Listmail Acct
• [Asterisk-Users] Compile error on Zaptel with Suse 9.1 (follow-up of subject: What is the best Linux for asterisk)   Johannes van Hulst
• [Asterisk-Users] New $89 VOIP phone   Robert Hajime Lanning
• [Asterisk-Users] New $89 VOIP phone   Steve Szmidt
• [Asterisk-Users] New $89 VOIP phone   Chris Shaw
• [Asterisk-Users] __use_ast_pthread_create_instead__   Chris Shaw
• [Asterisk-Users] Inter-digit timers on t100   Jason Kawakami
• [Asterisk-Users] Compile error on Zaptel with Suse 9.1 (follow-up of subject: What is the best Linux for asterisk)   Robert Lawrence
• [Asterisk-Users] New $89 VOIP phone   spectro
• [Asterisk-Users] couple basic questions   John Williams
• [Asterisk-Users] Compile error on Zaptel with Suse 9.1 (follow-up of subject: What is the best Linux for asterisk)   Dave Cotton
• [Asterisk-Users] couple basic questions   Chris Shaw
• [Asterisk-Users] couple basic questions   tmeeks at che-llp.com
• [Asterisk-Users] couple basic questions   Greg Hill
• [Asterisk-Users] Hunt Groups   Chris Modesitt
• [Asterisk-Users] New $89 VOIP phone   Derek Listmail Acct
• [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released   Wojciech Tryc
• [Asterisk-Users] MeetMe   Mitul Sen
• [Asterisk-Users] Dialplan problem - incoming calls get MOH, not ringing.   Patrick Lidstone (Personal e-mail)
• [Asterisk-Users] SpanDSP RxFax receives junk - gets Fax3Decode2d: Warning, (FakeInput): .... messages   Jon Bebeau
• [Asterisk-Users] BroadVOX   Mohammed Salim
• [Asterisk-Users] New $89 VOIP phone   Chris Shaw
• [Asterisk-Users] MeetMe   Gurdeep Singh Bagga Guru
• [Asterisk-Users] MeetMe   Gurdeep Singh Bagga Guru
• [Asterisk-Users] local echo using SPA-3000 as FXO port   Wolfgang S. Rupprecht
• [Asterisk-Users] MeetMe   Gurdeep Singh Bagga Guru
• [Asterisk-Users] MeetMe   Gurdeep Singh Bagga Guru
• [Asterisk-Users] Dialplan problem - incoming calls get MOH, not ringing.   Seth Remington
• [Asterisk-Users] New $89 VOIP phone   Derek Listmail Acct
• [Asterisk-Users] MeetMe   Steven Critchfield
• [Asterisk-Users] New $89 VOIP phone   Andrew Kohlsmith
• [Asterisk-Users] New $89 VOIP phone   Chris Shaw
• [Asterisk-Users] New $89 VOIP phone   Chris Shaw
• [Asterisk-Users] New $89 VOIP phone   Chris Shaw
• [Asterisk-Users] New $89 VOIP phone   Andrew Kohlsmith
• [Asterisk-Users] New $89 VOIP phone   Dee Lowndes
• [Asterisk-Users] Fax Detect over Analog   Ronan
• [Asterisk-Users] couple basic questions   Karlheinz Hagen
• [Asterisk-Users] dialplan woes   defiance
• [Asterisk-Users] dialplan woes   Steven Critchfield
• [Asterisk-Users] dialplan woes   Joshua McClintock
• [Asterisk-Users] dialplan woes   Andrew Kohlsmith
• [Asterisk-Users] dialplan woes   William Glynn
• [Asterisk-Users] dialplan woes   defiance
• [Asterisk-Users] dialplan woes   Steven Critchfield
• [Asterisk-Users] couple basic questions   james edwards
• [Asterisk-Users] multiple sound cards - howto have IP phone on each ?   Robert Rozman
• [Asterisk-Users] dialplan woes   defiance
• [Asterisk-Users] queue_log analysis   lenz
• [Asterisk-Users] couple basic questions   Tom Masterson
• [Asterisk-Users] couple basic questions   Chris Shaw
• [Asterisk-Users] couple basic questions   james edwards
• [Asterisk-Users] couple basic questions   Chris Shaw
• [Asterisk-Users] couple basic questions   Tom Masterson
• [Asterisk-Users] zaphfc in mode TE can't dialout (dialin is OK)   Colin Barschel
• [Asterisk-Users] Hunt Groups   Robert Jackson
• [Asterisk-Users] dialplan woes   Chris Shaw
• [Asterisk-Users] Agi Question   Yelson
• [Asterisk-Users] Hunt Groups   Chris Shaw
• [Asterisk-Users] Help - is voip good for in-house calls?   Francis Augusto Medeiros
• [Asterisk-Users] Help - is voip good for in-house calls?   Francis Augusto Medeiros
• [Asterisk-Users] MeetMe   Darryl Ross
• [Asterisk-Users] asterisk-wide variables   Darryl Ross
• [Asterisk-Users] Help - is voip good for in-house calls?   Darryl Ross
• [Asterisk-Users] asterisk and qsig   aleph lammim
• [Asterisk-Users] asterisk and qsig   Andrew Kohlsmith
• [Asterisk-Users] Agi Question   Steven Critchfield
• [Asterisk-Users] budgetone 101 and buttons   Michael George
• [Asterisk-Users] Soft DSS for Asterisk   Nicholas Bachmann
• [Asterisk-Users] queue_log analysis   Kevin Walsh
• [Asterisk-Users] [probably OT] wireless voip converter   Damjan
• [Asterisk-Users] 7960 help   Mark Woods
• [Asterisk-Users] Vlan question   Mark Woods
• [Asterisk-Users] Teliax TOS copied from Vonage?   Kevin Walsh
• [Asterisk-Users] [probably OT] wireless voip converter   Steven Critchfield
• [Asterisk-Users] 1.0 RC2 - FXS issues - no ring, no MWI   John Sellens
• [Asterisk-Users] Vlan question   Tenorio, Leandro
• [Asterisk-Users] Help   Dipak
• [Asterisk-Users] Avaya firmware   Tenorio, Leandro
• [Asterisk-Users] Internal extensions giving weird static-like dialtone   Lucas Wrenn
• [Asterisk-Users] Internal extensions giving weird static-like dialtone   Greg Hill
• [Asterisk-Users] [probably OT] wireless voip converter   Kevin Walsh
• [Asterisk-Users] Digium Hardware Question from Newbie   Jeff Borders
• [Asterisk-Users] New $89 VOIP phone   Jay Milk
• [Asterisk-Users] Asterisk and MEGACO   Arsen Chaloyan
• [Asterisk-Users] New $89 VOIP phone   Jay Milk
• [Asterisk-Users] Digium Hardware Question from Newbie   Michael Bielicki
• [Asterisk-Users] New $89 VOIP phone   Jay Milk
• [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released   Shaun Ewing
• [Asterisk-Users] Commercial deployments of Asterix   Mayank Mishra
• [Asterisk-Users] New $89 VOIP phone   Stefaan
• [Asterisk-Users] Help - is voip good for in-house calls?   Peter Svensson
• [Asterisk-Users] Help - is voip good for in-house calls?   Peter Svensson
• [Asterisk-Users] [probably OT] wireless voip converter   steve at daviesfam.org
• [Asterisk-Users] [probably OT] wireless voip converter   steve at daviesfam.org
• [Asterisk-Users] Commercial deployments of Asterix   Peter Svensson
• [Asterisk-Users] SIP providers USA   Sjaak Nabuurs
• [Asterisk-Users] Hangups - SIGFPE in dsp.c   Manfred Petz
• [Asterisk-Users] Festival Installation - Asterisk 1.0-RC2 && Debian Woody   Darryl Ross
• [Asterisk-Users] Festival Installation - Asterisk 1.0-RC2 && Debian Woody   Sebastian Sporleder
• [Asterisk-Users] Help - is voip good for in-house calls?   Holger Schurig
• [Asterisk-Users] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   clive18 at webmail.co.za
• [Asterisk-Users] Festival Installation - Asterisk 1.0-RC2 && Debian Woody   Darryl Ross
• [Asterisk-Users] Help - is voip good for in-house calls?   Holger Schurig
• [Asterisk-Users] Asterisk and MEGACO   Holger Schurig
• [Asterisk-Users] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   Ahmad Faiz
• [Asterisk-Users] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   Miroslav Nachev
• [Asterisk-Users] Re: Hangups - SIGFPE in dsp.c   Manfred Petz
• [Asterisk-Users] SIP providers USA   Kevin Walsh
• [Asterisk-Users] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   Miroslav Nachev
• [Asterisk-Users] Help   el Flynn
• [Asterisk-Users] How to accept the call and without billing the caller?   steven louse
• [Asterisk-Users] pickup any call   Altus Snyman
• [Asterisk-Users] SIP / IAX provider in the Netherlands.   micke at party.pp.se
• [Asterisk-Users] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   el Flynn
• [OT] RE: [Asterisk-Users] SIP / IAX provider in the Netherlands.   Andreas Sikkema
• [Asterisk-Users] Digium Hardware Question from Newbie   Kevin Walsh
• [Asterisk-Users] Audio problem   X
• [Asterisk-Users] Adtran power consumption   el Flynn
• [Asterisk-Users] New $89 VOIP phone   Holger Schurig
• [Asterisk-Users] SpanDSP   Simone Ricci
• [Asterisk-Users] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work   Miroslav Nachev
• [Asterisk-Users] New $89 VOIP phone   Holger Schurig
• [Asterisk-Users] DID Terminations   Andrew Yager
• [Asterisk-Users] RE: asterisk-wide variables   Kramer, R.D.J.
• [Asterisk-Users] Help Needed on these doubts   Mayank Mishra
• [Asterisk-Users] Channel bank for asterisk   Eran Gal
• [Asterisk-Users] Problems compiling chan_capi-0.3.5   Jason Williams
• [Asterisk-Users] RE: asterisk-wide variables   Leif Madsen
• [Asterisk-Users] Channel bank for asterisk   Andrew Kohlsmith
• [Asterisk-Users] pickup any call   Andrew Kohlsmith
• [Asterisk-Users] How to accept the call and without billing the caller?   Andrew Kohlsmith
• [Asterisk-Users] pickup any call   Altus Snyman
• [Asterisk-Users] DID Terminations   Olle E. Johansson
• [Asterisk-Users] External MW Lamp On/Off   Dunc
• [Asterisk-Users] pickup any call   Andrew Kohlsmith
• [Asterisk-Users] pickup any call   Altus Snyman
• [Asterisk-Users] pickup any call   Klaus-Peter Junghanns
• [Asterisk-Users] List-etiquette * AGAIN *   Olle E. Johansson
• [Asterisk-Users] asterisk start   Nicola Murino
• [Asterisk-Users] 7960 help   Donald Hall
• [Asterisk-Users] asterisk and qsig   aleph lammim
• [Asterisk-Users] Network Crashed when we try to start TDMoE   Miroslav Nachev
• [Asterisk-Users] How to accept the call and without billing the caller?   Christian Victor
• [Asterisk-Users] pickup any call   Andrew Kohlsmith
• [Asterisk-Users] call-back example   Maros RAJNOCH
• [Asterisk-Users] [OT] What's changing /etc/hosts?   Michael Welter
• [Asterisk-Users] pickup any call   Jeff Roberts
• [Asterisk-Users] Asterisk and Dial-Up ISP   Mark_C_Thomas at bankone.com
• [Asterisk-Users] pickup any call   Andrew Kohlsmith
• [Asterisk-Users] How to make RTP Packets NOT passing thru Asterisk?   Senthil Murugan
• [Asterisk-Users] Asterisk as SMS Service Center   Roland Zagler
• [Asterisk-Users] 7960 help   Shaun Ewing
• [Asterisk-Users] Choppiness/Ticking sounds over LAN   Richard Cook
• [Asterisk-Users] Question about TE405P   Angel Diaz
• [Asterisk-Users] How to make RTP Packets NOT passing thru Asterisk?   Simone Ricci
• [Asterisk-Users] Asterisk as SMS Service Center   Simone Ricci
• [Asterisk-Users] Question about TE405P   Bruno Fontana
• [Asterisk-Users] Question about TE405P   Andrew Kohlsmith
• [Asterisk-Users] Testing null values: ast_yyerror(): syntax error   Walt Reed
• [Asterisk-Users] Asterisk and Dial-Up ISP   Kevin Walsh
• [Asterisk-Users] asterisk start   administrator tootai
• [Asterisk-Users] [OT] What's changing /etc/hosts?   administrator tootai
• [Asterisk-Users] New $89 VOIP phone   Chris Shaw
• [Asterisk-Users] Asterisk as SMS Service Center   administrator tootai
• [Asterisk-Users] Asterisk and Dial-Up ISP   Johnathan Bunn
• [Asterisk-Users] Choppiness/Ticking sounds over LAN   Dana Nowell
• [Asterisk-Users] Avaya firmware   Aaron Johnson
• [Asterisk-Users] Mpg123 clarification   Olle E. Johansson
• [Asterisk-Users] Asterisk as SMS Service Center   Simone Ricci
• [Asterisk-Users] Config for sipgate?   Patrick
• [Asterisk-Users] paging/intercom   defiance
• [Asterisk-Users] Another Digium Hardware Question   John Bohman
• [Asterisk-Users] New $89 VOIP phone   robert.johnson-perkins at uk.nomura.com
• [Asterisk-Users] New $89 VOIP phone   Peter Svensson
• [Asterisk-Users] Pingtel and some chinese company   Mike Reed
• [Asterisk-Users] How to accept the call and without billing the caller?   Jeremy McNamara
• AW: [Asterisk-Users] Config for sipgate?   Markus Engelbrecht
• [Asterisk-Users] 7960 help   Mark Woods
• [Asterisk-Users] [OT] What's changing /etc/hosts?   Mark Woods
• AW: [Asterisk-Users] Problems compiling chan_capi-0.3.5   Markus Engelbrecht
• [Asterisk-Users] [OT] What's changing /etc/hosts?   Chris Shaw
• [Asterisk-Users] New $89 VOIP phone   Marcelo Pacheco
• [Asterisk-Users] [OT] What's changing /etc/hosts?   Andrew Kohlsmith
• [Asterisk-Users] paging/intercom   Jeff Roberts
• [Asterisk-Users] Hunt Groups   Chris A. Icide
• [Asterisk-Users] Problem compiling zaphfc   Christian Victor
• [Asterisk-Users] OT: New $89 VOIP phone   Robert Hajime Lanning
• [Asterisk-Users] spandsp   David Filion
• [Asterisk-Users] RE: New $85 VOIP Phone   Chris Shaw
• [Asterisk-Users] New $89 VOIP phone   asterisk at bsius.com
• [Asterisk-Users] Newbie physical layout question   Paul D. Warnagiris
• [Asterisk-Users] Newbie physical layout question   Marty Mastera
• [Asterisk-Users] my iaxy makes a strange noise...   Conrad Vermeulen
• [Asterisk-Users] Restart Digium Cards   Martin Keding
• [Asterisk-Users] Restart Digium Cards   Kanwar Ranbir Sandhu
• [Asterisk-Users] Restart Digium Cards   Martin Keding
• [Asterisk-Users] Restart Digium Cards   Dave Cotton
• [Asterisk-Users] Restart Digium Cards   Joseph
• [Asterisk-Users] queue_log analysis   Steve Szmidt
• [Asterisk-Users] service zaptel start   Christopher L. Wade
• [Asterisk-Users] SIP/IAX2 mysql auth + FreeBSD   Lerale Erwan
• [Asterisk-Users] Compile error on Zaptel with Suse 9.1 (follow-up of subject: What is the best Linux for asterisk)   Tony Nichols
• [Asterisk-Users] New $89 VOIP phone   Chris Shaw
• [Asterisk-Users] queue_log analysis   Richard Lyman
• [Asterisk-Users] Inter-digit timers on t100   Tony Nichols
• [Asterisk-Users] PCI Express and Digium Cards   Jorge Verastegui
• [Asterisk-Users] Inter-digit timers on t100   Kris Boutilier
• [Asterisk-Users] spandsp   Seth Remington
• [Asterisk-Users] spandsp   Jon Bebeau
• [Asterisk-Users] Color in console [SOLVED]   Dan Mahoney, System Admin
• [Asterisk-Users] Small patch to zaptel Makefile   John Morris
• [Asterisk-Users] Re: budgetone 101 and buttons   Stephen R. Besch
• [Asterisk-Users] queue_log analysis   Steve Szmidt
• [Asterisk-Users] [OT] What's changing /etc/hosts?   Mark Woods
• [Asterisk-Users] couple basic questions   John Williams
• [Asterisk-Users] Newbie Question   me peaceout
• [Asterisk-Users] Automated FWD configuration of Asterisk on OSX   Sunrise Ltd
• [Asterisk-Users] PCI Express and Digium Cards   Scott Laird
• [Asterisk-Users] Newbie Question   Joshua McClintock
• [Asterisk-Users] Newbie Question   Chris Shaw
• [Asterisk-Users] Newbie Question   Chris Shaw
• [Asterisk-Users] static noise TDM400P+T100P+Adtran Channel bank(TA750)   jong suk Lee
• [Asterisk-Users] Caller ID problem   Hall, Eric M.
• [Asterisk-Users] PCI 2.2 ??   Kelly Murphy
• [Asterisk-Users] Newbie Question   Andrew Kohlsmith
• [Asterisk-Users] Help! Cisco 7960 - dialing any digit makes the phone go nuts!   Joe Babstock
• [Asterisk-Users] Three tdm400p's (loaded with FXOs)   Ryan Courtnage
• [Asterisk-Users] Help! Cisco 7960 - dialing any digit makes the phone go nuts!   Chris Verges
• [Asterisk-Users] Help! Cisco 7960 - dialing any digit makes the phone go nuts!   Chris Shaw
• [Asterisk-Users] cisco phones w/ asterisk   Chris Verges
• [Asterisk-Users] SIPp and asterisk question   Tom Masterson
• [Asterisk-Users] couple basic questions   Roderick A. Anderson
• [Asterisk-Users] Three tdm400p's (loaded with FXOs)   Andrew Kohlsmith
• [Asterisk-Users] cisco phones w/ asterisk   Scott Laird
• [Asterisk-Users] Another small suggestion patch   John Morris
• [Asterisk-Users] How to accept the call and without billing the caller?   steven louse
• [Asterisk-Users] External MW Lamp On/Off   Greg Blakely
• [Asterisk-Users] cisco phones w/ asterisk   Lex Lethol
• [Asterisk-Users] CID on internal extensions   Ronan
• [Asterisk-Users] CID on internal extensions   Lyle Giese
• [Asterisk-Users] CID on internal extensions   Ronan
• [Asterisk-Users] CID on internal extensions   Bryan Vyhmeister
• [Asterisk-Users] CID on internal extensions   Lyle Giese
• [Asterisk-Users] Not a User Yet - but have some questions   Peter Harrison
• [Asterisk-Users] CID on internal extensions   Ronan
• [Asterisk-Users] Not a User Yet - but have some questions   jparr at bgcfreedom.com
• [Asterisk-Users] spandsp   Adam Goryachev
• [Asterisk-Users] CID on internal extensions   Darryl Ross
• [Asterisk-Users] CID on internal extensions   Ronan
• [Asterisk-Users] CID on internal extensions   Simon Brown
• [Asterisk-Users] queue name too long when sending sms over 32 chars   daniel schirrmacher
• [Asterisk-Users] queue name too long when sending sms over 32 chars   Stefan Reuter
• [Asterisk-Users] Re: Small patch to zaptel Makefile   Tony Mountifield
• [Asterisk-Users] Mpg123 clarification   Holger Schurig
• [Asterisk-Users] How to make RTP Packets NOT passing thru Asterisk?   Asterisk .
• [Asterisk-Users] RE: New $85 VOIP Phone   Holger Schurig
• [Asterisk-Users] Festival Issues   Darryl Ross
• [Asterisk-Users] service zaptel start   Dave Cotton
• [Asterisk-Users] Problem compiling zaphfc   Massimo De Nadal
• [Asterisk-Users] Pingtel and some chinese company   Dominique Kull
• [Asterisk-Users] queue_log analysis   lenz
• [Asterisk-Users] not yet a new user, some questions   Nicola Murino
• [Asterisk-Users] not yet a new user, some questions   Adam Goryachev
• [Asterisk-Users] call-back example   Tobias Jönsson
• [Asterisk-Users] queue name too long when sending sms over 32 chars   daniel schirrmacher
• [Asterisk-Users] queue_log analysis   lenz
• [Asterisk-Users] Asterisk & SER   JackSuroff
• [Asterisk-Users] queue_log analysis   Holger Schurig
• [Asterisk-Users] Zaptel module loading (was: Another small suggestion patch)   Tobias Jönsson
• [Asterisk-Users] queue_log analysis   administrator tootai
• [Asterisk-Users] call-back example   Peter Svensson
• [Asterisk-Users] Error message "phone_mini_packet Read returned -1" and strange Internet PhoneCard behavior   Jim O'Brien
• [Asterisk-Users] Floating point exception help   Gary Pigott
• [Asterisk-Users] queue_log analysis   Adam Goryachev
• [Asterisk-Users] Multiple SIP phones ringing for same extension   David Gurr
• [Asterisk-Users] Floating point exception help   Manfred Petz
• [Asterisk-Users] couple basic questions   Peter Corlett
• [Asterisk-Users] Multiple SIP phones ringing for same extension   Peter Svensson
• [Asterisk-Users] chan_phone - compilation problem   Marcin Mazurek
• [Asterisk-Users] call-back example   Tobias Jönsson
• [Asterisk-Users] chan_phone - compilation problem   Dave Cotton
• [Asterisk-Users] call-back example   Peter Svensson
• [Asterisk-Users] Floating point exception help   Gary Pigott
• [Asterisk-Users] SIP reinvite code negotiation   Andreas Sikkema
• [Asterisk-Users] Using Asterisk as a voicemail only with legacy switch   Brian Hudson
• [Asterisk-Users] asterisk + ser   Pavel Siderov
• [Asterisk-Users] fax output from Asterisk into file   Vladyslav
• [Asterisk-Users] Multiple SIP phones ringing for same extension   Nicolas Gudino
• [Asterisk-Users] Inbound IAX2 calls has no music on hold   Steve Szmidt
• [Asterisk-Users] Echo SIP-T100P-PRI   Mike Schwartz
• [Asterisk-Users] Echo SIP-T100P-PRI   Peter Svensson
• [Asterisk-Users] Inter-digit timers on t100   Tony Nichols
• [Asterisk-Users] asterisk + ser   Miroslav Nachev
• [Asterisk-Users] Floating point exception help   Gary Pigott
• [Asterisk-Users] spandsp   Seth Remington
• [Asterisk-Users] spandsp   Seth Remington
• [Asterisk-Users] Floating point exception help   Gary Pigott
• [Asterisk-Users] queue_log analysis   lenz
• [Asterisk-Users] spandsp   Simone Ricci
• [Asterisk-Users] Re: spandsp   David Filion
• [Asterisk-Users] Problem compiling zaphfc   Christian Victor
• [Asterisk-Users] Re: Echo SIP-T100P-PRI   Mike Schwartz
• [Asterisk-Users] Problem compiling zaphfc   Christian Victor
• [Asterisk-Users] Re: Echo SIP-T100P-PRI   Peter Svensson
• [Asterisk-Users] spandsp   William Glynn
• [Asterisk-Users] Re: Echo SIP-T100P-PRI   Jorge Mendoza
• [Asterisk-Users] T100P PCI or PCI-x   Christopher Jacob
• [Asterisk-Users] spandsp   administrator tootai
• [Asterisk-Users] spandsp   Jon Bebeau
• [Asterisk-Users] spandsp   Simone Ricci
• [Asterisk-Users] spandsp   Simone Ricci
• [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?   James Freire
• [Asterisk-Users] spandsp   Simone Ricci
• [Asterisk-Users] T100P PCI or PCI-x   mattf
• [Asterisk-Users] SpanDSP/RxFax help...   Rob Fugina
• [Asterisk-Users] spandsp   William Glynn
• [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?   Walt Reed
• [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?   Jay Milk
• [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?   Chris Lee
• [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?   James Freire
• [Asterisk-Users] No Success with SwissVoice.   Matthew Boehm
• [Asterisk-Users] Three tdm400p's (loaded with FXOs)   Ryan Courtnage
• [Asterisk-Users] Debit/Credit Card Terminals   Trevor Peirce
• [Asterisk-Users] Problem compiling zaphfc   Massimo De Nadal
• [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?   Ryan Courtnage
• [Asterisk-Users] T100P PCI or PCI-x   Steven Critchfield
• [Asterisk-Users] Debit/Credit Card Terminals   Andrew Kohlsmith
• [Asterisk-Users] No Success with SwissVoice.   Florian Overkamp
• [Asterisk-Users] Dial from AGI [MSG]   Morgan Gilroy
• [Asterisk-Users] Three tdm400p's (loaded with FXOs)   Andrew Kohlsmith
• [Asterisk-Users] Andre Bierwirth's ring state patches for SNOM 200 programable buttons   David Hinkle
• [Asterisk-Users] Problems loading chan_h323 on Opteron 64 bit   Roger Schreiter
• [Asterisk-Users] Dial from AGI [MSG]   Steven Critchfield
• [Asterisk-Users] Three tdm400p's (loaded with FXOs)   Scott Laird
• [Asterisk-Users] Success with SwissVoice.   Matthew Boehm
• [Asterisk-Users] Problem compiling zaphfc   Christian Victor
• [Asterisk-Users] Opencall.org and SpandDSP   Muiz Motani
• [Asterisk-Users] UserAgent support for MGCP   Girouard, Marc
• [Asterisk-Users] Opencall.org and SpandDSP   Scott Laird
• [Asterisk-Users] Isdn4Linux and DTMF   Shaun Ewing
• [Asterisk-Users] Debit/Credit Card Terminals   Trevor Peirce
• [Asterisk-Users] Compile problem   steveb at asbc.demon.co.uk
• [Asterisk-Users] Compile problem   Steven Critchfield
• [Asterisk-Users] Opencall.org and SpandDSP   Muiz Motani
• [Asterisk-Users] Pingtel registration failing   Anton Yurchenko
• [Asterisk-Users] Matthias Urlichs: Urlaub/Vacation   smurf at server.smurf.noris.de
• [Asterisk-Users] Opencall.org and SpandDSP   Brian McManus
• [Asterisk-Users] Does Granstream BT100 Conference Button Work?   James Freire
• Searchable Archive (was:Re: [Asterisk-Users] Opencall.org and SpandDSP)   Muiz Motani
• [Asterisk-Users] Re: Searchable Archive   Bryan Vyhmeister
• [Asterisk-Users] Re: Pingtel registration failing   Tobias Jönsson
• [Asterisk-Users] Dial from AGI [MSG]   Morgan Gilroy
• [Asterisk-Users] Compile problem   steveb at asbc.demon.co.uk
• [Asterisk-Users] Does Granstream BT100 Conference Button Work?   Chris Shaw
• [Asterisk-Users] Re: Searchable Archive   Klaus Darilion
• [Asterisk-Users] Does Granstream BT100 Conference Button Work?   James Freire
• [Asterisk-Users] Avaya firmware   Brian Elton
• [Asterisk-Users] False Hangups on Asterisk   Ruben Fagundo
• [Asterisk-Users] Does Granstream BT100 Conference Button Work?   Chris Shaw
• [Asterisk-Users] Re: Searchable Archive   James H. Thompson
• [Asterisk-Users] Does Granstream BT100 Conference Button Work?   Steve Szmidt
• [Asterisk-Users] Does Granstream BT100 Conference Button Work?   Chris Shaw
• [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?   Andrew Thompson
• [Asterisk-Users] Granstream BT100 Rings Once and Waits for Call Pickup?   Kanuri, Seshu
• [Asterisk-Users] spandsp   administrator tootai
• [Asterisk-Users] Granstream BT100 Rings Once and Waits for Call Pickup?   Chris Shaw
• [Asterisk-Users] Inband announcement of parking slot from app _parkandannounce?   Kris Boutilier
• [Asterisk-Users] Granstream BT100 Rings Once and Waits for Call Pickup?   Chris Shaw
• [Asterisk-Users] Inband announcement of parking slot from app_parkandannounce?   Chris Shaw
• [Asterisk-Users] More on Broadvox   Dan Mahoney, System Admin
• [Asterisk-Users] More on Broadvox   Andrew Kohlsmith
• [Asterisk-Users] Received packet with bad UDP checksum   Mike Benoit
• [Asterisk-Users] residential sip phone   John Williams
• [Asterisk-Users] How to run different codecs between the same endpoints on an IAX trunk?   Kris Boutilier
• [Asterisk-Users] GrandStream BT101 Attended Transfers   Chris Shaw
• [Asterisk-Users] More on Broadvox   Dan Mahoney, System Admin
• [Asterisk-Users] More on Broadvox   Dan Mahoney, System Admin
• [Asterisk-Users] Re: Echo SIP-T100P-PRI   Rich Adamson
• [Asterisk-Users] Alternative SIP phone   Jean-Yves Avenard
• [Asterisk-Users] How to run different codecs between the same endpoints on an IAX trunk?   Chris Shaw
• [Asterisk-Users] GrandStream BT101 Attended Transfers   Ryan Courtnage
• [Asterisk-Users] More on Broadvox   Andrew Kohlsmith
• [Asterisk-Users] More on Broadvox   Andrew Kohlsmith
• [Asterisk-Users] AGI Script: calleridnamelookup.agi   Greg Blakely
• [Asterisk-Users] More on Broadvox   Chris Shaw
• [Asterisk-Users] GrandStream BT101 Attended Transfers   Chris Shaw
• [Asterisk-Users] cisco phones w/ asterisk   Rich Adamson
• [Asterisk-Users] ATA with Built-in Switch   Marty Mastera
• [Asterisk-Users] GrandStream BT101 Attended Transfers   Ryan Courtnage
• [Asterisk-Users] residential sip phone   Tony Richards
• [Asterisk-Users] residential sip phone   Chris Shaw
• [Asterisk-Users] GrandStream BT101 Attended Transfers   Chris Shaw
• [Asterisk-Users] More on Broadvox   Rich Adamson
• [Asterisk-Users] Request for help designing an unusual * application   Lee Allen
• [Asterisk-Users] GrandStream BT101 Attended Transfers   Chris Shaw
• [Asterisk-Users] Re: Searchable Archive   Muiz Motani
• [Asterisk-Users] CID on internal extensions   Ronan
• [Asterisk-Users] Inbound IAX2 calls has no music on hold   Robert Barnes
• [Asterisk-Users] How to run different codecs between the same endpoints on an IAX trunk?   William Glynn
• [Asterisk-Users] IAX2 Port strangeness   Jeremy Bogan
• [Asterisk-Users] More on Broadvox   James Cloos
• [Asterisk-Users] Request for help designing an unusual * application   Jay Milk
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   Matt Darnell
• [Asterisk-Users] Request for help designing an unusual * application   Adam Goryachev
• [Asterisk-Users] IAX2 Port strangeness   Andres
• [Asterisk-Users] IAX2 Port strangeness   Jeremy Bogan
• [Asterisk-Users] CallerId   Simon Brown
• [Asterisk-Users] CallerId   Simon Brown
• [Asterisk-Users] CallerId   Darryl Ross
• [Asterisk-Users] Re: Isdn4Linux and DTMF   Shaun Ewing
• [Asterisk-Users] CallerId   Simon Brown
• [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)   Shaun Ewing
• [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)   Kimble Young
• [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)   Greg Smith
• [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)   Simon Brown
• [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)   Adam Goryachev
• [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)   Matthew Enger
• [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)   Shaun Ewing
• [Asterisk-Users] Atick Certification on FXO Modules (Australia)   Christopher Lee
• [Asterisk-Users] Atick Certification on FXO Modules (Australia)   Simon Brown
• [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)   Darryl Ross
• [Asterisk-Users] How to run different codecs between the same endpoints on an IAX trunk?   steve at daviesfam.org
• [Asterisk-Users] Success with SwissVoice.   Florian Overkamp
• [Asterisk-Users] Does Granstream BT100 Conference Button Work?   Shaun Ewing
• [Asterisk-Users] Atick Certification on FXO Modules (Australia)   Paul Liew
• [Asterisk-Users] Re: Searchable Archive   Olle E. Johansson
• [Asterisk-Users] False Hangups on Asterisk   Vassilis Konstantinou
• [Asterisk-Users] from Newcomtech Co,. Ltd Help us.   maya mn
• [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)   David MacKinnon
• [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)   Jean-Yves Avenard
• [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)   David MacKinnon
• [Asterisk-Users] Operator-type phone   el Flynn
• [Asterisk-Users] spandsp   Simone Ricci
• [Asterisk-Users] GrandStream BT101 Attended Transfers   Massimo De Nadal
• [Asterisk-Users] spandsp   administrator tootai
• [Asterisk-Users] spandsp   Simone Ricci
• [Asterisk-Users] telnet and Root   neil
• [Asterisk-Users] problem in with mysql modules.conf to load cdr_addon_mysql.so   DIPAK PAUL
• [Asterisk-Users] Number unavailable   Thomas Kuepper
• [Asterisk-Users] spandsp   administrator tootai
• [Asterisk-Users] Problem with asterisk and pgsql   DIPAK PAUL
• [Asterisk-Users] telnet and Root   Thomas Kuepper
• [Asterisk-Users] Re: telnet and Root   Roberto Piola
• [Asterisk-Users] x100p won't answer   Imran Akbar
• [Asterisk-Users] Multi-bitrate codecs   Simone Ricci
• [Asterisk-Users] IAXY S100I noise   Conrad Vermeulen
• [Asterisk-Users] problem in with mysql modules.conf to load cdr_addon_mysql.so   Lerale Erwan
• [Asterisk-Users] Problem in with mysql modules.conf to load cdr_addon_mysql.so   DIPAK PAUL
• [Asterisk-Users] Problem in with mysql modules.conf to load cdr_addon_mysql.so   Lerale Erwan
• [Asterisk-Users] How to run different codecs between the same endpoints on an IAX trunk?   Andrew Kohlsmith
• [Asterisk-Users] How to run different codecs between the same endpoints on an IAX trunk?   Andrew Kohlsmith
• [Asterisk-Users] Request for help designing an unusual * application   Rich Adamson
• [Asterisk-Users] Zaptel Problem after Upgrade   Kurt Bauer
• [Asterisk-Users] Finding operator from ISDN signalling?   Roy Sigurd Karlsbakk
• [Asterisk-Users] help   Kevin VanBorrendam
• [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?   James Freire
• [Asterisk-Users] Sipura endpoints   Matt Schulte
• [Asterisk-Users] help   Dave Cotton
• [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?   James Freire
• [Asterisk-Users] Request for help designing an unusual * application   Lee Allen
• [Asterisk-Users] telnet and Root   Steve Szmidt
• [Asterisk-Users] help (OT)   Andrew Thompson
• [Asterisk-Users] Sipura endpoints   Tim McKee
• [Asterisk-Users] Creating 79xx Configs   Joseph
• [Asterisk-Users] Request for help designing an unusual *application   Lee Allen
• [Asterisk-Users] Request for help designing an unusual * application   Lee Allen
• [Asterisk-Users] False Hangups on Asterisk   matt.riddell at sineapps.com
• [Asterisk-Users] Help - is voip good for in-house calls?   Francis Augusto Medeiros
• [Asterisk-Users] DID Questions   Kanuri, Seshu
• [Asterisk-Users] Request for help designing an unusual * application   Andrew Thompson
• [Asterisk-Users] False Hangups on Asterisk   Ruben Fagundo
• [Asterisk-Users] dual servers   Altus Snyman
• [Asterisk-Users] spandsp   William Glynn
• [Asterisk-Users] How to run different codecs between the same endpoints on an IAX trunk?   William Glynn
• [Asterisk-Users] Re: problem in with mysql modules.conf to load cdr_addon_mysql.so (Lerale Erwan)   Walter Klomp
• [Asterisk-Users] tT funktions   Thomas Kuepper
• [Asterisk-Users] tT funktions   Eric Wieling
• [Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?   Walt Reed
• [Asterisk-Users] tT funktions   Dominique Kull
• [Asterisk-Users] Does Granstream BT100 Conference Button Work?   Eric Wieling
• [Asterisk-Users] from Newcomtech Co,. Ltd Help us.   luan
• [Asterisk-Users] telnet and Root   Chris Shaw
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   James Freire
• [Asterisk-Users] telnet and Root   Walt Reed
• [Asterisk-Users] telnet and Root   Chris Shaw
• [Asterisk-Users] Sipura endpoints   Andres
• [Asterisk-Users] Handling invalid extensions   Craig Guy
• [Asterisk-Users] Alternative SIP phone   Ariel's Hotmail
• [Asterisk-Users] Creating 79xx Configs   Ben Merrills
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   Craig Guy
• [Asterisk-Users] dual servers   Brian McManus
• [Asterisk-Users] Handling invalid extensions   William Glynn
• [Asterisk-Users] Creating 79xx Configs   Joseph
• [Asterisk-Users] Stream File (AGI) question   Ben Merrills
• [Asterisk-Users] Handling invalid extensions   William Glynn
• [Asterisk-Users] Creating 79xx Configs   Joseph Finley
• [Asterisk-Users] Creating 79xx Configs   Ben Merrills
• [Asterisk-Users] telnet and Root   Steve Szmidt
• [Asterisk-Users] Creating 79xx Configs   Brian McManus
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   Chris Shaw
• [Asterisk-Users] telnet and Root   Steven Critchfield
• [Asterisk-Users] how to collect user entered digits   John Millican
• [Asterisk-Users] Inbound IAX2 calls has no music on hold   Michael Graves
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   James Freire
• [Asterisk-Users] telnet and Root   Steve Szmidt
• [Asterisk-Users] Creating 79xx Configs   Joseph
• [Asterisk-Users] Inbound IAX2 calls has no music on hold   Steve Szmidt
• [Asterisk-Users] telnet and Root   Walt Reed
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   Walt Reed
• [Asterisk-Users] Oh323 installing problem make[1]: *** [asteriskaudio.o] Error 1   Diego CHILE
• [Asterisk-Users] how to collect user entered digits   Walt Reed
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   Chris Shaw
• [Asterisk-Users] telnet and Root   Ejay Hire
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   James Freire
• [Asterisk-Users] telnet and Root   Chris Shaw
• [Asterisk-Users] Sipura endpoints   Trevor Peirce
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   Chris Shaw
• [Asterisk-Users] Inbound IAX2 calls has no music on hold   Eric Wieling
• [Asterisk-Users] how to collect user entered digits   John Millican
• [Asterisk-Users] TDM FXO or TE410P codec translation problem   JackyChen
• [Asterisk-Users] Incoming MSN via ZapHFC -> to SIP   Bastian Schern
• [Asterisk-Users] Sipura endpoints   Mike Benoit
• [Asterisk-Users] Floating point exception help   Manfred Petz
• [Asterisk-Users] chan_h323 doesn't pass audio before call is answered   Michael Ulitskiy
• [Asterisk-Users] CDR problems with MySQL   Chris HARIGA
• [Asterisk-Users] determining what number was dialed?   Paul Concepcion
• [Asterisk-Users] determining what number was dialed?   Steven Critchfield
• [Asterisk-Users] determining what number was dialed?   Richard Cook
• [Asterisk-Users] Silence on incoming monitored calls   Nick Cobley
• [Asterisk-Users] Silence on incoming monitored calls   Steven Critchfield
• [Asterisk-Users] Help with upgrading 7960 SCCP to SIP   Kevin Day
• [Asterisk-Users] CDR problems with MySQL   Malcolm Bader
• [Asterisk-Users] CDR problems with MySQL   Chris HARIGA
• [Asterisk-Users] Is posissble TE405P ?   Angel Diaz
• [Asterisk-Users] Help with upgrading 7960 SCCP to SIP   Jerimiah Cole
• [Asterisk-Users] determining what number was dialed?   Chris Shaw
• [Asterisk-Users] CDR problems with MySQL   Joshua McClintock
• [Asterisk-Users] determining what number was dialed?   Steven Critchfield
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   John Baker
• [Asterisk-Users] TE410P - ZT_CHANCONFIG failed   Arnaud Pignard
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   Olle E. Johansson
• [Asterisk-Users] Help with upgrading 7960 SCCP to SIP   Kevin Day
• [Asterisk-Users] determining what number was dialed?   Chris Shaw
• [Asterisk-Users] TE410P - ZT_CHANCONFIG failed   Peter Svensson
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   Chris Shaw
• [Asterisk-Users] determining what number was dialed?   Chris Shaw
• [Asterisk-Users] Sipura endpoints   Nathan C. Smith
• [Asterisk-Users] CDR problems with MySQL   Chris HARIGA
• [Asterisk-Users] Sipura endpoints   Chris Shaw
• [Asterisk-Users] determining what number was dialed?   Steven Critchfield
• [Asterisk-Users] TE410P - ZT_CHANCONFIG failed   Arnaud Pignard
• [Asterisk-Users] Invalid module format   Michael George
• [Asterisk-Users] Help with upgrading 7960 SCCP to SIP   Simon Brown
• [Asterisk-Users] Three tdm400p's (loaded with FXOs)   Kevin P. Fleming
• [Asterisk-Users] Help with upgrading 7960 SCCP to SIP   Doug Shubert
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   Robert Rozman
• [Asterisk-Users] snom 200 and * question   listas iPfone
• [Asterisk-Users] Help with upgrading 7960 SCCP to SIP   Matt Darnell
• [Asterisk-Users] Max retries exceeded on call - seqno 102 (critical request)   Steve Anderson
• [Asterisk-Users] Help with upgrading 7960 SCCP to SIP   Rich Adamson
• [Asterisk-Users] x100p won't answer   Lyle Giese
• [Asterisk-Users] BT Communicator (SIP???) and Asterisk   Robert Boardman
• [Asterisk-Users] Testing a channel's status   Luke Cyca
• [Asterisk-Users] Strange problem with Dial   Michael George
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   Chris Shaw
• [Asterisk-Users] Adding macros causes ringing to fail   Chris Shaw
• [Asterisk-Users] Adding macros causes ringing to fail   Chris Shaw
• [Asterisk-Users] Strange problem with Dial   Michael George
• [Asterisk-Users] telnet and Root   Mark Woods
• [Asterisk-Users] Help with upgrading 7960 SCCP to SIP   Mark Woods
• [Asterisk-Users] Forwarding PSTN to PSTN?   F. Edward Barrett
• [Asterisk-Users] Testing a channel's status   John Todd
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   Craig Guy
• [Asterisk-Users] Pulse dialing...   John Todd
• [Asterisk-Users] Handling invalid extensions   Craig Guy
• [Asterisk-Users] Forwarding PSTN to PSTN?   Greg Broiles
• [Asterisk-Users] Sipura partners with Linksys for new combo router/SIP ATA   James H. Thompson
• [Asterisk-Users] Asterisk PBX Functions via SIP phone   Karl Brose
• [Asterisk-Users] spandsp tiff debian question   Adrian Serafini
• [Asterisk-Users] PoE injectors   Gonzalo Gasca Meza
• [Asterisk-Users] PoE injectors   Shaun Ewing
• [Asterisk-Users] Incoming MSN via ZapHFC -> to SIP   Massimo De Nadal
• [Asterisk-Users] TE410P - ZT_CHANCONFIG failed   Martin List-Petersen
• [Asterisk-Users] hangup question   Ilia Mirkin
• [Asterisk-Users] just-added second X100P   spectro
• [Asterisk-Users] Sipura endpoints   spectro
• [Asterisk-Users] telnet and Root   neil
• [Asterisk-Users] Asterisk and AVM FritzBox Fon (Germany)   Marco Czudej
• [Asterisk-Users] system reboot often?   Michael George
• [Asterisk-Users] system reboot often?   Steven Critchfield
• [Asterisk-Users] welltech fxo and *   Danny Zak
• [Asterisk-Users] system reboot often?   Lyle Giese
• [Asterisk-Users] zaptel config   Imran Akbar
• [Asterisk-Users] zaptel config   Lyle Giese
• [Asterisk-Users] Problems with festival   Sergio Serrano
• [Asterisk-Users] just-added second X100P   Mike Benoit
• [Asterisk-Users] Finding operator from ISDN signalling?   Tobias Jönsson
• [Asterisk-Users] Zultys 4x4 or 4x5   Michael Graves
• [Asterisk-Users] autocreatepeer and sip peer options   yair at hakak.com
• [Asterisk-Users] residential sip phone   Daniel Jimenez
• [Asterisk-Users] How to minimally configure modules.conf loading?   Bryce Nesbitt (mailing list account)
• [Asterisk-Users] Multi-bitrate codecs   Kevin Walsh
• [Asterisk-Users] residential sip phone   programming dept
• [Asterisk-Users] Testing null values: ast_yyerror(): syntax error   Grzegorz Nosek
• [Asterisk-Users] Asterisk and software Raid   Ed Devine
• [Asterisk-Users] asterisk in india   Eran Gal
• [Asterisk-Users] Asterisk and software Raid   Lyle Giese
• [Asterisk-Users] Asterisk and software Raid   Michael Welter
• [Asterisk-Users] cmd Monitor creating sound notification on channel   Jason Kawakami
• [Asterisk-Users] Asterisk and software Raid   Greg Boehnlein
• [Asterisk-Users] Asterisk and software Raid   Greg Boehnlein
• [Asterisk-Users] residential sip phone   William Suffill
• [Asterisk-Users] Asterisk and software Raid   Rémi Letot
• [Asterisk-Users] Can I place calls to arbitrary sip uri address from windows messenger 5 ?   Robert Rozman
• [Asterisk-Users] Number and name for SIP extension at the same time ?   Robert Rozman
• [Asterisk-Users] ActXPhone Active X control link is dead - has anyone cached files ?   Robert Rozman
• [Asterisk-Users] determining what number was dialed?   Paul Concepcion
• [Asterisk-Users] Queue Monitor   Kevin
• [Asterisk-Users] Asterisk and software Raid   Andrew Kohlsmith
• [Asterisk-Users] determining what number was dialed?   Kevin P. Fleming
• [Asterisk-Users] Asterisk and software Raid   Andrew Kohlsmith
• [Asterisk-Users] determining what number was dialed?   Paul Concepcion
• [Asterisk-Users] determining what number was dialed?   Paul Concepcion
• [Asterisk-Users] Asterisk and software Raid   Steve Szmidt
• [Asterisk-Users] Inbound IAX2 calls has no music on hold   Steve Szmidt
• [Asterisk-Users] Asterisk and software Raid   niles at atheos.net
• [Asterisk-Users] need help with zaptel.   Edward Huitt
• [Asterisk-Users] need help with zaptel.   Greg Blakely
• [Asterisk-Users] Cisco IP Phone- disjoin conference   Assaf Benharoosh
• [Asterisk-Users] IAX2 DTMF not recognized - Bug report - Help sought   Bryce Nesbitt (mailing list account)
• [Asterisk-Users] need help with zaptel.   Edward Huitt
• [Asterisk-Users] Uniden UIP200 Review   Tim // NCS
• [Asterisk-Users] Zultys 4x4 or 4x5   Duane
• [Asterisk-Users] Sipura partners with Linksys for new combo router/SIP ATA   Jay Milk
• [Asterisk-Users] Asterisk and software Raid   Chris
• [Asterisk-Users] Sipura partners with Linksys for new combo router/SIP ATA   Kevin Walsh
• [Asterisk-Users] Spandsp - opencall.org offline   Roland Zagler
• [Asterisk-Users] Spandsp - opencall.org offline   Seth Remington
• [Asterisk-Users] BT Communicator (SIP???) and Asterisk   gARetH baBB
• [Asterisk-Users] app_mp3 with bri-stuff.0.1.0RC4a does not work   Deti Fliegl
• [Asterisk-Users] Sipura partners with Linksys for new combo router/SIP ATA   John Todd
• [Asterisk-Users] tsu 120   John R. Hill
• [Asterisk-Users] Release 1.0 of FWD Assistant for MacOSX now available   Sunrise Ltd
• [Asterisk-Users] We have thousands of U ISDN interface phones   Marcelo Pacheco
• [Asterisk-Users] just-added second X100P   spectro
• [Asterisk-Users] skinny or sccp?   Pavel Jezek
• [Asterisk-Users] asterisk T100P to Merlin Legend   spectro
• [Asterisk-Users] skinny or sccp?   Julien Goodwin
• [Asterisk-Users] Re: SpanDSP/RxFax help...   Stefan Tichy
• [Asterisk-Users] How to limit asterisk to use only three channels   Bartosz Wegrzyn
• [Asterisk-Users] system reboot often?   Michael George
• [Asterisk-Users] Error compiling meetme2   Geoff Nordli
• [Asterisk-Users] IAX2 DTMF not recognized - Bug report - Help sought   Marty Mastera
• [Asterisk-Users] How to limit asterisk to use only three channels   Senad Jordanovic
• [Asterisk-Users] debian install zaptel   Sergio Serrano
• [Asterisk-Users] Re: SpanDSP/RxFax help...   Carlos Hernandez
• [Asterisk-Users] MusicOnHold problem   Robert Rozman
• [Asterisk-Users] Uniden UIP200 Review   Ryan Courtnage
• [Asterisk-Users] Re: SpanDSP/RxFax help...   William Suffill
• [Asterisk-Users] Error compiling meetme2   Geoff Nordli
• [Asterisk-Users] Re: SpanDSP/RxFax help...   Carlos Hernandez
• [Asterisk-Users] SIP Phone recommendation for Receptionist   el Flynn
• [Asterisk-Users] Pulse dialed digit recognization   Daniel Bichara
• [Asterisk-Users] Queue Calls without using the   Michael at Matraex.com
• [Asterisk-Users] Pulse dialed digit recognization   Shaun Ewing
• [Asterisk-Users] determining what number was dialed?   Steven Critchfield
• [Asterisk-Users] SIP Phone recommendation for Receptionist   Jeremy Bogan
• [Asterisk-Users] SIP Phone recommendation for Receptionist   Jeremy McNamara
• [Asterisk-Users] ActXPhone Active X control link is dead - has anyone cached files ?   Jorge Cisneros Flores
• [Asterisk-Users] SIP Phone recommendation for Receptionist   James H. Thompson
• [Asterisk-Users] Queue Calls without using the   Adam Goryachev
• [Asterisk-Users] zap show channels - no such command   Imran Akbar
• [Asterisk-Users] zap show channels - no such command   Jeremy McNamara
• [Asterisk-Users] zap show channels - no such command   el Flynn
• [Asterisk-Users] zap show channels - no such command   Imran Akbar
• [Asterisk-Users] zap show channels - no such command   Jeremy McNamara
• [Asterisk-Users] zap show channels - no such command   Imran Akbar
• [Asterisk-Users] zap show channels - no such command   Jon Radon
• [Asterisk-Users] zap show channels - no such command   Imran Akbar
• [Asterisk-Users] zap show channels - no such command   el Flynn
• [Asterisk-Users] zap show channels - no such command   Imran Akbar
• [Asterisk-Users] zap show channels - no such command   Adam Goryachev
• [Asterisk-Users] zap show channels - no such command   Darryl Ross
• [Asterisk-Users] zap show channels - no such command   Imran Akbar
• [Asterisk-Users] zap show channels - no such command   Imran Akbar
• [Asterisk-Users] zap show channels - no such command   Imran Akbar
• [Asterisk-Users] Problem with mysql and with asterisk   DIPAK PAUL
• [Asterisk-Users] Problem with mysql and with asterisk   Adam Hart
• [Asterisk-Users] Bug in recording uavarible   dome at tel.co.th
• [Asterisk-Users] Problem with asterisk and postgresql   DIPAK PAUL
• [Asterisk-Users] 2 servers   Altus Snyman
• [Asterisk-Users] Choosing between TE405P and TE410P   Tony Mountifield
• [Asterisk-Users] False Hangups on Asterisk   Florian Overkamp
• [Asterisk-Users] using ChanIsAvail   Poul Pedersen
• [Asterisk-Users] MusicOnHold problem   Chris Stenton
• [Asterisk-Users] How to recover the problem with pgsql and asterisk   DIPAK PAUL
• [Asterisk-Users] Adit 600 FXO & FXS   Vasyl Rublyov
• [Asterisk-Users] Adit 600 FXO & FXS   Greg Smith
• [Asterisk-Users] Choosing between TE405P and TE410P   Scott Stingel
• [Asterisk-Users] Re: 2 servers   David Cook
• [Asterisk-Users] Question about dial out via Zap   Hall, Eric M.
• [Asterisk-Users] Re: Choosing between TE405P and TE410P   Tony Mountifield
• [Asterisk-Users] Adit 600 FXO & FXS   Vasyl Rublyov
• [Asterisk-Users] zaptel installation   Imran Akbar
• [Asterisk-Users] zap show channels - no such command   Brent Franks
• [Asterisk-Users] SIP Phone recommendation for Receptionist   Paul Mahler
• [Asterisk-Users] routing telephone calls via "switchboard/asterisk".   Stig Thune
• [Asterisk-Users] Multi-bitrate codecs   Steve Kann
• [Asterisk-Users] routing telephone calls via "switchboard/asterisk".   Scott Stingel
• [Asterisk-Users] asterisk T100P to Merlin Legend   Steven Sokol
• [Asterisk-Users] Cisco 7940 Question   Christopher L. Wade
• [Asterisk-Users] HFC-S in NT mode, wiring?   Simone Ricci
• [Asterisk-Users] routing telephone calls via"switchboard/asterisk".   Stig Thune
• [Asterisk-Users] extensions.conf   Steve Maroney
• [Asterisk-Users] zaptel installation   Michael George
• [Asterisk-Users] Swissvoice MGCP Error 502   Matthew Boehm
• [Asterisk-Users] SIP / IAX provider in the Netherlands.   Edwig Knol
• [Asterisk-Users] SIP "unphones"   Jay Milk
• [Asterisk-Users] Question about dial out via Zap   Hall, Eric M.
• [Asterisk-Users] telnet and Root   Chris Travers
• [Asterisk-Users] telnet and Root   Chris Travers
• [Asterisk-Users] telnet and Root   Chris Travers
• [Asterisk-Users] HFC-S in NT mode, wiring?   Bodo Hahnke
• [Asterisk-Users] using ChanIsAvail   Chris Shaw
• [Asterisk-Users] determining what number was dialed?   Chris Shaw
• [Asterisk-Users] SIP "unphones"   listas iPfone
• [Asterisk-Users] MGCP and dialing out   Matthew Boehm
• [Asterisk-Users] extensions.conf   Steven Critchfield
• [Asterisk-Users] Cisco 7940 Question   Shaun Ewing
• [Asterisk-Users] (no subject)   David Cook
• [Asterisk-Users] SIP Phone recommendation for Receptionist   Kanuri, Seshu
• [Asterisk-Users] Error compiling meetme2   Steve Luzynski
• [Asterisk-Users] SIP "unphones"   Brian McSpadden
• [Asterisk-Users] Re: skinny or sccp?   Pavel Jezek
• [Asterisk-Users] Guide to getting started with Asterisk on MacOSX   Sunrise Ltd
• [Asterisk-Users] BT Communicator (SIP???) and Asterisk   Robert Boardman
• [Asterisk-Users] Queue Monitor   Robert Jackson
• [Asterisk-Users] Question about dial out via Zap   Lyle Giese
• [Asterisk-Users] Re: 2 servers   Kanuri, Seshu
• [Asterisk-Users] Problem with asterisk and postgresql   Lyle Giese
• [Asterisk-Users] asterisk in india   Kanuri, Seshu
• [Asterisk-Users] Cisco 7940 Question   jparr at bgcfreedom.com
• [Asterisk-Users] asterisk T100P to Merlin Legend   Vasyl Rublyov
• [Asterisk-Users] Uniden UIP200 Review   spectro
• [Asterisk-Users] Asterisk Install on Kernel 2.6.x   Shawn Parker
• [Asterisk-Users] SIP "unphones"   James H. Thompson
• [Asterisk-Users] Re: skinny or sccp?   Pavel Jezek
• [Asterisk-Users] Firmware Update IAXy   BetaTeilchen
• [Asterisk-Users] Re: HFC-S in NT mode, wiring?   Stefan Tichy
• [Asterisk-Users] extensions.conf   Michael George
• [Asterisk-Users] SIP "unphones"   Chris Shaw
• [Asterisk-Users] Cisco 7960G, Skinny.conf, and reboots   Matthew Boehm
• [Asterisk-Users] Strange problem with Dial   Michael George
• [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)   Roy Sigurd Karlsbakk
• [Asterisk-Users] Asterisk <------- Quintum SIP Registration   Krystian Filiks
• [Asterisk-Users] multiple X100P config?   spectro
• [Asterisk-Users] Asterisk and software Raid   William Boehlke
• [Asterisk-Users] Asterisk WITH Swyx... Any Idea?   Zineddin Karzazi
• [Asterisk-Users] VoicePluse DID problem   dome at tel.co.th
• [Asterisk-Users] newb question regarding DTMF   Erik Anderson
• [Asterisk-Users] DONE ! Firmware Update IAXy   BetaTeilchen
• [Asterisk-Users] Error compiling meetme2   Geoff Nordli
• [Asterisk-Users] SIP "unphones"   Chris Shaw
• [Asterisk-Users] RE: SIP "unphones"   Chris Shaw
• [Asterisk-Users] Hold the phone!   Roderick A. Anderson
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Scott Stingel
• [Asterisk-Users] Cisco 7960G, Skinny.conf, and reboots   Jeremy McNamara
• [Asterisk-Users] SIP "unphones"   Jay Milk
• [Asterisk-Users] Error compiling meetme2   Geoff Nordli
• [Asterisk-Users] Asterisk Install on Kernel 2.6.x   Lubomir Christov
• [Asterisk-Users] Uniden UIP200 Review   Ryan Courtnage
• [Asterisk-Users] SIP "unphones"   Jay Milk
• [Asterisk-Users] How to make RTP Packets NOT passing thru Asterisk?   Partha Sarathi
• [Asterisk-Users] (no subject)   Ryan Courtnage
• [Asterisk-Users] spandsp make error mmx.h   Adrian Serafini
• [Asterisk-Users] Re: 2 servers   Nguyen Quang Hoa
• [Asterisk-Users] H323 outgoing calls   Darren Wiebe
• [Asterisk-Users] Telemarketer screening   david kwok
• [Asterisk-Users] strange problem PBX-Asterisk   Kurt Bauer
• [Asterisk-Users] Asterisk Install on Kernel 2.6.x   Apollon Koutlides
• [Asterisk-Users] Polycom SoundPoint IP 500/600 XML minibrowser   Matt Darnell
• [Asterisk-Users] Asterisk Install on Kernel 2.6.x   Trevor G. Hammonds
• [Asterisk-Users] Asterisk Install on Kernel 2.6.x   Maciej Kietlinski
• [Asterisk-Users] g729 Codec on Max OS X   Darryl Ross
• [Asterisk-Users] iax to iax link   Raul Elizondo (wizardteam)
• [Asterisk-Users] Inband DTMF is not supported on codec G.711 u-law. Use RFC2833   asterisk at bortal.de
• [Asterisk-Users] hfc-s card, brii-stuff.0.1.0-RC4a, zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled.   Maurizio Marini
• [Asterisk-Users] h323 direct calls   Thomas Kuepper
• [Asterisk-Users] RE: Re: 2 servers   David Cook
• [Asterisk-Users] determining what number was dialed?   Paul Concepcion
• [Asterisk-Users] If you run Archlinux...   Shawn Parker
• [Asterisk-Users] Cisco 7960G, Skinny.conf, and reboots   Kubat, Philip
• [Asterisk-Users] Bell Canada Caller-ID   Matt G
• [Asterisk-Users] Sangoma Card Support   Isamar Maia
• [Asterisk-Users] Asterisk <------- Quintum SIP Registration   Matthew Boehm
• [Asterisk-Users] Problem with sound on Wildcard TE410P   Claus Futtrup
• [Asterisk-Users] MGCP 1.0 NCS 1.0 on a motorola SBV4200   Kiss Karoly
• [Asterisk-Users] Asterisk MIBS   Gary Carr
• [Asterisk-Users] SIP Provider in India/Pakistan/Bengladesh   Cesar Hernandez
• [Asterisk-Users] problems with the mailing list??   Bodo Hahnke
• [Asterisk-Users] Re: newb question regarding DTMF   Erik Anderson
• [Asterisk-Users] Asterisk to Vonage   Paterson, Mark
• [Asterisk-Users] VoicePluse DID problem   Marcelo Pacheco
• [Asterisk-Users] Asterisk Install on Kernel 2.6.x   Steve Szmidt
• [Asterisk-Users] Hold the phone!   Richard Cook
• [Asterisk-Users] newb question regarding DTMF   Reid A. Forrest
• [Asterisk-Users] Asterisk WITH Swyx... Any Idea?   Steven Critchfield
• [Asterisk-Users] MGCP problem   Kiss Karoly
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Bob Knight
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Steven Critchfield
• [Asterisk-Users] SIP "unphones"   Chris Shaw
• [Asterisk-Users] Inband DTMF is not supported on codec G.711 u-law. Use RFC2833   Steven Critchfield
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Reid A. Forrest
• [Asterisk-Users] Swissvoice IP10S and RTP Port Operation   Matthew Boehm
• [Asterisk-Users] Autoattend detecting same digit twice   klussier at comcast.net
• [Asterisk-Users] Re: newb question regarding DTMF   Chris Shaw
• [Asterisk-Users] [Asterisk-Dev] Asterisks   Huddleston, Robert
• [Asterisk-Users] Telemarketer screening   Steve Szmidt
• [Asterisk-Users] Error compiling meetme2   Nicolas Gudino
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Michael Welter
• [Asterisk-Users] call queue help   defiance
• [Asterisk-Users] R X100P connected to Meridan-1 system will notdisconnect call   Reid A. Forrest
• [Asterisk-Users] newb question regarding DTMF   Greg Hill
• [Asterisk-Users] Voicemail & "Couldn't read username" error   Bill
• [Asterisk-Users] Warning when I use iax2 for inbound and outbound calls   Ariel's Hotmail
• [Asterisk-Users] H323 outgoing calls   Daniel Bichara
• [Asterisk-Users] Bell Canada Caller-ID   Richard Cook
• [Asterisk-Users] Asterisk Install on Kernel 2.6.x   Marcelo Pacheco
• [Asterisk-Users] Uniden UIP200 Review   Ryan Courtnage
• [Asterisk-Users] Help with upgrading 7960 SCCP to SIP   Kevin Day
• [Asterisk-Users] Asterisk Install on Kernel 2.6.x   mattf
• [Asterisk-Users] strange problem PBX-Asterisk   Christian Victor
• [Asterisk-Users] ex-girlfriend logic not working in latest CVS?   James Sizemore
• [Asterisk-Users] CVS RPMs for Mandrake 10 (Zaptel and Asterisk)   Scott Petersen
• [Asterisk-Users] Hold the phone!   Andrew Thompson
• [Asterisk-Users] newb question regarding DTMF   Erik Anderson
• [Asterisk-Users] Bell Canada Caller-ID   Andrew Kohlsmith
• [Asterisk-Users] sip to sip calls thru asterisk   Gary Carr
• [Asterisk-Users] determining what number was dialed?   Steven Critchfield
• [Asterisk-Users] Astricon - call for help   joachim
• [Asterisk-Users] Asterisk Install on Kernel 2.6.x   Marcelo Pacheco
• [Asterisk-Users] ex-girlfriend logic not working in latest CVS?   Greg Hill
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Steve Szmidt
• [Asterisk-Users] ex-girlfriend logic not working in latest CVS?   Josh Roberson
• [Asterisk-Users] Broken Pipe   Mike Roberts
• [Asterisk-Users] Hold the phone!   Roderick A. Anderson
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Scott Stingel
• [Asterisk-Users] Inband DTMF is not supported on codec G.711 u-law. Use RFC2833   Eric Wieling
• [Asterisk-Users] Autoattend detecting same digit twice   Peter Svensson
• [Asterisk-Users] R X100P connected to Meridan-1 system will not disconnect call   Mark
• [Asterisk-Users] Re: [Asterisk-Dev] Asterisks   Greg Hill
• [Asterisk-Users] MailMan slowness...   Chris Shaw
• [Asterisk-Users] CPU utilization   Michael Welter
• [Asterisk-Users] Re: Voicemail & "Couldn't read username" error   Bill
• [Asterisk-Users] Telemarketer screening   Walt Reed
• [Asterisk-Users] Trouble with all linux sip softphones.... And asterisk/linphone/kphone SRPMs   John Morris
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Benjamin Johnson
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Peter Svensson
• [Asterisk-Users] Monitor() hangs   Manuel Wenger
• [Asterisk-Users] Directed Call Pickup   Mike Meyer
• [Asterisk-Users] Asterisk WITH Swyx... Any Idea?   Zineddin Karzazi
• [Asterisk-Users] call queue help   Andrew Thompson
• [Asterisk-Users] Bell Canada Caller-ID   Kanwar Ranbir Sandhu
• [Asterisk-Users] SMP Performance   Tim Jackson
• [Asterisk-Users] X100P connected to Meridan-1 system will not disconnect call   Mark Farquharson
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Gary Carr
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Tim Jackson
• [Asterisk-Users] strange problem PBX-Asterisk   Peter Svensson
• [Asterisk-Users] how to collect user entered digits   John Millican
• [Asterisk-Users] Error compiling meetme2   Geoff Nordli
• [Asterisk-Users] Re: Voicemail & "Couldn't read username" error   Eric Wieling
• [Asterisk-Users] Euro isdn caller id problem - SOLVED   Asterisk
• [Asterisk-Users] Using Lucent/Avaya 64XX sets with asterisk   Peter Miller
• [Asterisk-Users] VoicePluse DID problem   Jeremy Bogan
• [Asterisk-Users] Problem with Asterisk VM + MySQL Database   TSS Support
• [Asterisk-Users] off list   Chris Clifton
• [Asterisk-Users] ex-girlfriend logic not working in latest CVS?   James Sizemore
• [Asterisk-Users] MailMan slowness...   mastapasta
• [Asterisk-Users] Inband DTMF is not supported on codec G.711 u-law. Use RFC2833   Linterra
• [Asterisk-Users] Inband DTMF is not supported on codec G.711 u-law. Use RFC2833   Steven Critchfield
• [Asterisk-Users] call queue help   defiance
• [Asterisk-Users] Asterisk with Adit 600   Craig Neumanns
• [Asterisk-Users] Voicemail & "Couldn't read username" error   Umar Sear
• [Asterisk-Users] Re: newb question regarding DTMF   Umar Sear
• [Asterisk-Users] RC2 and Netmeeting 3.01 ?   Robert Rozman
• [Asterisk-Users] Remotely change call forward   Russell Horn
• [Asterisk-Users] Asterisk with Adit 600   Chris Shaw
• [Asterisk-Users] Broken Pipe   Scott Petersen
• [Asterisk-Users] Asterisk with Adit 600   Michael Welter
• [Asterisk-Users] Inband DTMF is not supported on codec G.711u-law. Use RFC2833   Chris Shaw
• [Asterisk-Users] Parking and Extensions   Matthew Boehm
• [Asterisk-Users] MusicOnHold problem   Robert Rozman
• [Asterisk-Users] RE: Parking and Extensions   Matthew Boehm
• [Asterisk-Users] [Asterisk Users] Help with SIP Hosted Billing Service   E Samuels
• [Asterisk-Users] Remotely change call forward   Ryan Courtnage
• [Asterisk-Users] Remotely change call forward   Chris Shaw
• [Asterisk-Users] Grandstream Budgetone BT-101 and VoipJet   John Week
• [Asterisk-Users] Asterisk to Vonage   Jay Milk
• [Asterisk-Users] SMP Performance   Steve Szmidt
• [Asterisk-Users] SIP "unphones"   Jay Milk
• [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones   Andrew Elchuk
• [Asterisk-Users] Asterisk to Vonage   Chris Shaw
• [Asterisk-Users] Grandstream Budgetone BT-101 and VoipJet   William Suffill
• [Asterisk-Users] SIP "unphones"   Adam Goryachev
• [Asterisk-Users] SIP "unphones"   William Suffill
• [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones   William Suffill
• [Asterisk-Users] SIP "unphones"   Chris Shaw
• [Asterisk-Users] Asterisk to Vonage   Paterson, Mark
• [Asterisk-Users] Grandstream Budgetone BT-101 and VoipJet   Glenn Dalgliesh
• [Asterisk-Users] Grandstream Budgetone BT-101 and VoipJet   Chris Shaw
• [Asterisk-Users] RC2 and Netmeeting 3.01 ?   Zineddin Karzazi
• [Asterisk-Users] X100P connected to Meridan-1 system will not disconnect call   Leo Ann Boon
• [Asterisk-Users] Perl AGI - no output from agi script to Asterisk   Robert Rozman
• [Asterisk-Users] Can I have same numbers in different contexts ?   Robert Rozman
• [Asterisk-Users] MusicOnHold problem   Robert Rozman
• [Asterisk-Users] system reboot often?   Michael George
• [Asterisk-Users] How can i configure extensions.conf.   maya mn
• [Asterisk-Users] system reboot often?   Steven Critchfield
• [Asterisk-Users] sip to sip calls thru asterisk   Karl Brose
• [Asterisk-Users] Asterisk and software Raid   Greg Boehnlein
• [Asterisk-Users] Can I have same numbers in different contexts ?   Steven Critchfield
• [Asterisk-Users] system reboot often?   el Flynn
• [Asterisk-Users] Asterisk and software Raid   Chris
• [Asterisk-Users] system reboot often?   Scott Stingel
• [Asterisk-Users] system reboot often?   Steven Critchfield
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Craig Guy
• [Asterisk-Users] Zaptel/Zapata and SIP relationship   Steve D.
• [Asterisk-Users] how to collect user entered digits   John Millican
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Scott Stingel
• [Asterisk-Users] newb question regarding DTMF   Greg Hill
• [Asterisk-Users] Asterisk to Vonage   Greg Hill
• [Asterisk-Users] Asterisk to Vonage   Assaf Benharoosh
• [Asterisk-Users] Perl AGI - no output from agi script to Aste risk   mattf
• [Asterisk-Users] Asterisk to Vonage   Richard Cook
• [Asterisk-Users] Voicepulse incoming / dial extension   Anil Purohit
• [Asterisk-Users] RTP and NAT   WPJ mail
• [Asterisk-Users] how to collect user entered digits   Ryan Courtnage
• [Asterisk-Users] Dell PowerEdge 750 rackmount   Craig Guy
• [Asterisk-Users] desparate for help DEV LITE KIT   steve
• [Asterisk-Users] desparate for help DEV LITE KIT   Darryl Ross
• [Asterisk-Users] desparate for help DEV LITE KIT   Sean Cheesman
• [Asterisk-Users] SMP Performance   joachim
• [Asterisk-Users] Asterisk to Vonage   Jay Milk
• [Asterisk-Users] Closing bug reports without fixing the reported problem   Rob Wise
• [Asterisk-Users] Asterisk to Vonage   Jay Milk
• [Asterisk-Users] SIP "unphones"   Jay Milk
• [Asterisk-Users] asterisk directory service   aster esk
• [Asterisk-Users] Swissvoice IP10S and RTP Port Operation   Florian Overkamp
• [Asterisk-Users] strange problem PBX-Asterisk   Kurt Bauer
• [Asterisk-Users] Perl AGI - no output from agi script to Asterisk   Adam Goryachev
• [Asterisk-Users] Re: 2 servers   altus
• [Asterisk-Users] Closing bug reports without fixing the repor ted problem   Steve Hanselman
• [Asterisk-Users] Closing bug reports without fixing the repor ted problem   Peter Svensson
• [Asterisk-Users] Re: Cisco 7960G, Skinny.conf, and reboots   Pavel Jezek
• [Asterisk-Users] Compressing a dialplan   Tobias Jönsson
• [Asterisk-Users] Asterisk with Adit 600   Brent Franks
• [Asterisk-Users] Re: SIP Provider in India/Pakistan/Bengladesh   Atif Rasheed
• [Asterisk-Users] Parallel T1 cable?   el Flynn
• [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines   Steve Underwood
• [Asterisk-Users] Blocking a channel on T1   Sudhir Kumar
• [Asterisk-Users] GSM to BRI ISDN Gateway   Miroslav Nachev
• [Asterisk-Users] Loading module chan_oh323.so   Balgansuren Batsukh
• [Asterisk-Users] channel banks   Eran Gal
• [Asterisk-Users] sip history - which file ?   Stig Thune
• [Asterisk-Users] system reboot often?   Michael George
• [Asterisk-Users] Individual call-forwarding on ISDN   Stefan Neufeind
• [Asterisk-Users] freebsd 4.10 and port misc/zaptel   Dinesh Nair
• [Asterisk-Users] GSM to BRI ISDN Gateway   Nana Yaw
• [Asterisk-Users] GSM to BRI ISDN Gateway   Chris Lee
• [Asterisk-Users] Individual call-forwarding on ISDN   Peter Svensson
• [Asterisk-Users] GSM to BRI ISDN Gateway   Steve Kennedy
• [Asterisk-Users] Advice on BT ISDN Services (UK)   Benjamin Johnson
• [Asterisk-Users] Individual call-forwarding on ISDN   Stefan Neufeind
• [Asterisk-Users] Advice on BT ISDN Services (UK)   Jon Fautley
• [Asterisk-Users] GSM to BRI ISDN Gateway   Anton Tinchev
• [Asterisk-Users] GSM to BRI ISDN Gateway   Anton Tinchev
• [Asterisk-Users] desparate for help DEV LITE KIT   Ed Brady
• [Asterisk-Users] Cisco 7940 Question   Brian Pavane
• [Asterisk-Users] Asterisks   Huddleston, Robert
• [Asterisk-Users] Re: Compressing a dialplan   Maron Kristófersson
• [Asterisk-Users] system reboot often?   Rich Adamson
• [Asterisk-Users] system reboot often?   Steven Critchfield
• [Asterisk-Users] Advice on BT ISDN Services (UK)   Benjamin Johnson
• [Asterisk-Users] Re: Compressing a dialplan   Tobias Jönsson
• [Asterisk-Users] ericsson drg22 - will not connect to asterisk   Stig Thune
• [Asterisk-Users] Re: Compressing a dialplan   Leif Madsen
• [Asterisk-Users] Parallel T1 cable?   Steven Critchfield
• [Asterisk-Users] Incorrect sound   Johannes Morgenroth
• [Asterisk-Users] Advice on BT ISDN Services (UK)   Jon Fautley
• [Asterisk-Users] Blocking a channel on T1   Steven Critchfield
• [Asterisk-Users] Voicepulse incoming / dial extension   Andrew Thompson
• [Asterisk-Users] SMP Performance   Matt Schulte
• [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing linesand 20 phones   Andrew Thompson
• [Asterisk-Users] How to be taken out of the queue?   Kenny Lam
• [Asterisk-Users] Individual call-forwarding on ISDN   Peter Svensson
• [Asterisk-Users] Problem of set up asterisk-1.0-RC2.tar.gz with asterisk-prepaid-0.3.1   DIPAK PAUL
• [Asterisk-Users] SMP Performance   Matt Schulte
• [Asterisk-Users] SMP Performance   mattf
• [Asterisk-Users] SMP Performance   Tim Jackson
• [Asterisk-Users] Blocking a channel on T1   Christian Victor
• [Asterisk-Users] External BRI - is there such thing?   Francis Augusto Medeiros
• [Asterisk-Users] Asterisk PBX and backup Circuits   Kurt Pasewaldt
• [Asterisk-Users] KS-Like controlling   Francis Augusto Medeiros
• [Asterisk-Users] Blocking a channel on T1   Andrew Kohlsmith
• [Asterisk-Users] SMP Performance   joachim
• [Asterisk-Users] sip to sip calls thru asterisk   Gary Carr
• [Asterisk-Users] Problem of set up asterisk-1.0-RC2.tar.gz withasterisk-prepaid-0.3.1   Richard Cook
• [Asterisk-Users] TDM400P lockups (FXO)   Alok K. Dhir
• [Asterisk-Users] SMP Performance   mattf
• [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)   Dave Covert (Sailtech)
• [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)   Huddleston, Robert
• [Asterisk-Users] Asterisk PBX and backup Circuits   kurt x
• [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)   Scott Stingel
• [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)   Benjamin Johnson
• [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)   Huddleston, Robert
• [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)   Dave Covert (Sailtech)
• [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones   Andrew Elchuk
• [Asterisk-Users] chan_sccp2 & 7960 -- documentation and example request.   Robert Lawrence
• [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones   Jeff Roberts
• [Asterisk-Users] *, Sipura, and cisco dtmf   AJ Grinnell
• [Asterisk-Users] GrandStream HT-486 ATA as VoIP Gateway   Miroslav Nachev
• [Asterisk-Users] Re: how to collect user entered digits   Stephen R. Besch
• [Asterisk-Users] chan_oh323 and cdr   Roger Schreiter
• [Asterisk-Users] oh323_indicate: Ignoring PROGRESS indication   Roger Schreiter
• [Asterisk-Users] Re: how to collect user entered digits   John Millican
• [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones   Andrew Elchuk
• [Asterisk-Users] chan_sccp2 & 7960 -- documentation and examplerequest.   Craig Guy
• [Asterisk-Users] echo cancellation   Peter Ang
• [Asterisk-Users] TDM400P lockups (FXO)   Marty Mastera
• [Asterisk-Users] WTS: Just arrived Brand New Cisco IP phones   Jeffrey Richey
• [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones   spectro
• [Asterisk-Users] Sound deformation during conversation or IVR message   Régis MARTIN
• [Asterisk-Users] 2x HFC ISDN Cards - SuSE 9.1 - Problems with making calls   Marco Czudej
• [Asterisk-Users] Asterisk WITH Swyx... Any Idea?   Zineddin Karzazi
• [Asterisk-Users] Avaya dialing problems   Aaron Johnson
• [Asterisk-Users] Uniden UIP200 Review   spectro
• [Asterisk-Users] Sound deformation during conversation or IVR message   Peter Svensson
• [Asterisk-Users] channel banks   Joshua McClintock
• [Asterisk-Users] Uniden UIP200 Review   Devon Stephens
• [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones   Scott Laird
• [Asterisk-Users] Read and dtmf ?   Danny Zak
• [Asterisk-Users] Something broken in voicemail app??   Ferrara, Jamie
• [Asterisk-Users] Uniden UIP200 Review   Ryan Courtnage
• [Asterisk-Users] Something broken in voicemail app??   Brian McManus
• [Asterisk-Users] External BRI - is there such thing?   Jorge Mendoza
• [Asterisk-Users] Which end hungup?   steve at nexusuk.org
• [Asterisk-Users] WTS: Just arrived Brand New Cisco IP phones   Brent Franks
• [Asterisk-Users] Something broken in voicemail app??   Ferrara, Jamie
• [Asterisk-Users] Something broken in voicemail app??   Brian McManus
• [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones   James H. Thompson
• [Asterisk-Users] Something broken in voicemail app??   Brian McManus
• [Asterisk-Users] [Asterisk Users] Help with SIP Hosted Billing Service   Umar Sear
• [Asterisk-Users] chan_oh323: __use_ast_pthread_create_instead__ (was: chan_oh323 loading error)   Roger Schreiter
• [Asterisk-Users] Re: Compressing a dialplan   Maron Kristófersson
• [Asterisk-Users] Avaya dialing problems (w/ SIP debugging)   Aaron Johnson
• [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones   Nicolas Gudino
• [Asterisk-Users] Sound deformation during conversation or IVRmessage   Régis MARTIN
• [Asterisk-Users] Re: Parking and Extensions   Mike Meyer
• [Asterisk-Users] asterisk & chan_sccp   Chad Whitten
• [Asterisk-Users] Voicemail forwarding from SER & extensions.conf   VCI Help Desk
• [Asterisk-Users] asterisk & chan_sccp   Matthew Boehm
• [Asterisk-Users] Voip phones & headsets   neil
• [Asterisk-Users] Something broken in voicemail app??   Ferrara, Jamie
• [Asterisk-Users] chan_sccp with multi-lines and 7960's   Matthew Boehm
• [Asterisk-Users] unknown SMS Messages   Martin Holler
• [Asterisk-Users] Something broken in voicemail app??   Brian McManus
• [Asterisk-Users] Avaya dialing problems   Aaron Johnson
• [Asterisk-Users] asterisk & chan_sccp   Chad Whitten
• [Asterisk-Users] Distinctive Ring Cadences   Mike Meyer
• [Asterisk-Users] Avaya dialing problems   Aaron Johnson
• [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines   reseaux
• [Asterisk-Users] Uniden UIP200 Review   spectro
• [Asterisk-Users] spamdsp capacity   James H. Thompson
• [Asterisk-Users] Distinctive Ring Cadences   Jay Milk
• [Asterisk-Users] Distinctive Ring Cadences   Chris Shaw
• [Asterisk-Users] Uniden UIP200 Review   Terry Wilson
• [Asterisk-Users] New wiki pages: Avaya 4602 upgrade and configuration   Aaron Johnson
• [Asterisk-Users] 7960 Looses DHCP Lease when 7920 boots!?   justin at vergeworks.com
• [Asterisk-Users] Distinctive Ring Cadences   matt.riddell at sineapps.com
• [Asterisk-Users] Distinctive Ring Cadences   Chris Shaw
• [Asterisk-Users] Distinctive Ring Cadences   Jay Milk
• [Asterisk-Users] Error when closing asterisk   Tom Masterson
• [Asterisk-Users] Faxing with SPANDSP or any other mean ? Is it possible ? Am I dreaming ?   Jean-François Rousseau
• [Asterisk-Users] Something broken in voicemail app??   Ferrara, Jamie
• [Asterisk-Users] Distinctive Ring Cadences   Nicolas Gudino
• [Asterisk-Users] Distinctive Ring Cadences   matt.riddell at sineapps.com
• [Asterisk-Users] Fax detect   Simon Brown
• [Asterisk-Users] Voicemail forwarding from SER & extensions.conf   Leo Ann Boon
• [Asterisk-Users] IP Phone Recommendation ?   Gary G. Hendershot
• [Asterisk-Users] Phone recommendations   Jeff Borders
• [Asterisk-Users] chan_oh323: __use_ast_pthread_create_instead__ (was: chan_oh323 loading error)   Balgansuren Batsukh
• [Asterisk-Users] TDM400P lockups (FXO)   Michael George
• [Asterisk-Users] Fax detect   Ryan Courtnage
• [Asterisk-Users] IP Phone Recommendation ?   Ryan Courtnage
• [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones   Greg Broiles
• [Asterisk-Users] Fax detect   Nathan Wharton
• [Asterisk-Users] IP Phone Recommendation ?   Greg Broiles
• [Asterisk-Users] home lab setup   Mikel King
• [Asterisk-Users] Fax detect   Simon Brown
• [Asterisk-Users] IP Phone Recommendation ?   Jay Milk
• [Asterisk-Users] Voicepulse incoming / dial extension   Anil Purohit
• [Asterisk-Users] chan_sccp2 & 7960 -- documentation and example request.   Paterson, Mark
• [Asterisk-Users] chan_sccp2 & 7960 -- documentation andexample request.   Tim McKee
• [Asterisk-Users] Could use some advice/reality check from someone knowledgeable   Gary G. Hendershot
• [Asterisk-Users] Dial by name across 2 systems   Jim O'Brien
• [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines   Steve Underwood
• [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines   Steve Underwood
• [Asterisk-Users] Error Compiling MySQL Friends   imail
• [Asterisk-Users] Error Compiling MySQL Friends   el Flynn
• [Asterisk-Users] Re: Compressing a dialplan   Tobias Jönsson
• [Asterisk-Users] GrandStream HT-486 ATA as VoIP Gateway   Klaus Darilion
• [Asterisk-Users] Voip phones & headsets   Dave Cotton
• [Asterisk-Users] GrandStream HT-486 ATA as VoIP Gateway   Tobias Jönsson
• [Asterisk-Users] Parallel T1 cable?   el Flynn
• [Asterisk-Users] Voip phones & headsets   Shaun Ewing
• [Asterisk-Users] Distinctive Ring Cadences   William Suffill
• [Asterisk-Users] Out Dial Problem   R Wong
• [Asterisk-Users] system reboot often?   Richard Scobie
• [Asterisk-Users] I have on isdn phone and one isdn-card (ast-pbx) attached (to my line)...   Nicolas
• [Asterisk-Users] Re: 2x HFC ISDN Cards - SuSE 9.1 - Problems with making calls   Nicolas
• [Asterisk-Users] Codec   altus
• [Asterisk-Users] RE: SMP Performance   Nicolas
• [Asterisk-Users] I have on isdn phone and one isdn-card (ast-pbx)attached (to my line)...   Nicolas
• [Asterisk-Users] Grandstream Firmware   WipeOut
• [Asterisk-Users] GSM to BRI ISDN Gateway   Nana Yaw
• [Asterisk-Users] Grandstream Firmware   Duane
• [Asterisk-Users] Grandstream Firmware   WipeOut
• R: [Asterisk-Users] Grandstream Firmware   Manuel Wenger
• [Asterisk-Users] GSM to BRI ISDN Gateway   Steve Underwood
• [Asterisk-Users] TDM400P + 1FXS + 1FXO for sale   asterisk at anime.net
• [Asterisk-Users] GSM to BRI ISDN Gateway   Nana Yaw
• [Asterisk-Users] "for Lack of RTP activity in 0 seconds"   markus monka
• [Asterisk-Users] Faxing with SPANDSP or any other mean ? Is it possible ? Am I dreaming ?   Steve Underwood
• [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines   reseaux
• [Asterisk-Users] GSM to BRI ISDN Gateway   Steve Underwood
• [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines   Steve Underwood
• [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines   Vladyslav
• [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines   reseaux
• [Asterisk-Users] GSM to BRI ISDN Gateway   Steve Kennedy
• [Asterisk-Users] Noise on ZAP channel   Matt
• [Asterisk-Users] Asterisk media problem behind NAT   Partha Sarathi
• [Asterisk-Users] Noise on ZAP channel   Chris Glover
• [Asterisk-Users] chan_capi module   asterisk at bortal.de
• [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines   Steve Underwood
• [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines   reseaux
• [Asterisk-Users] Noise on ZAP channel   Matt
• [Asterisk-Users] Overhead Paging   Brian Pavane
• [Asterisk-Users] system reboot often?   Leif Madsen
• [Asterisk-Users] chan_capi module   Fabian Stelzer
• [Asterisk-Users] Re: Noise on ZAP channel   Tony Mountifield
• [Asterisk-Users] chan_capi module   asterisk at bortal.de
• [Asterisk-Users] need help with zaptel.   Johannes van Hulst
• [Asterisk-Users] chan_oh323: __use_ast_pthread_create_instead __ (was: chan_oh323 loading error)   Huddleston, Robert
• [Asterisk-Users] <<< AGI and EXEC function CONFIRMATION >>>   jeff quade
• [Asterisk-Users] GRSecurity and ALSA on a Gentoo Server   Deon Rodden
• [Asterisk-Users] Advice on BT ISDN Services (UK)   Jason Williams
• [Asterisk-Users] Can't make zaptel on red hat 9   PHP Mechanic
• [Asterisk-Users] Re: Overhead Paging   David Cook
• [Asterisk-Users] Which end hungup?   Jason Williams
• [Asterisk-Users] chan_oh323 build (resubmit w/ new title)   Huddleston, Robert
• [Asterisk-Users] ISDN Card Recommendation   Paterson, Mark
• [Asterisk-Users] Overhead Paging   Lyle Giese
• [Asterisk-Users] system reboot often?   Leif Madsen
• [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines   Jean-François Rousseau
• [Asterisk-Users] Codec   Andrew Kohlsmith
• [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines   Andrew Kohlsmith
• [Asterisk-Users] ISDN Card Recommendation   Fares Gianluca
• [Asterisk-Users] chan_sccp2 & 7960 -- documentation and example request.   Matthew Boehm
• Problems compiling chan_h323 (was: [Asterisk-Users] chan_oh323: __use_ast_pthread_create_instead ...)   Roger Schreiter
• [Asterisk-Users] spandsp and certain (e.g. Canon) fax machines   Tony Nichols
• [Asterisk-Users] Asterisk PBX and backup Circuits   Rich Adamson
• [Asterisk-Users] ISDN Card Recommendation   Rob Fugina
• [Asterisk-Users] Astricon hotel recommendations.....?   Nick Barnes
• [Asterisk-Users] Which end hungup?   steve at nexusuk.org
• [Asterisk-Users] Overhead Paging   Rich Adamson
• [Asterisk-Users] Which end hungup?   steve at nexusuk.org
• [Asterisk-Users] Astricon hotel recommendations.....?   Nick Barnes
• [Asterisk-Users] chan_sccp2 & 7960 -- documentation and example request.   Jan Czmok
• [Asterisk-Users] Astricon hotel recommendations.....?   Rick L. Wilson, Sr.
• [Asterisk-Users] Wil spandsp work with i4l driver?   Marcin Mazurek
• [Asterisk-Users] ilbc asterisk and handytone/Xpro   palsingh at blueyonder.co.uk
• [Asterisk-Users] chan_oh323 and cdr   Michael Manousos
• [Asterisk-Users] Asterisk+IVR functions trouble   alex3377
• [Asterisk-Users] RC2 and Netmeeting 3.01 ?   Michael Manousos
• [Asterisk-Users] Asterisk and Eicon Diva 2.0   Fares Gianluca
• [Asterisk-Users] asterisk-wide variables   Michael Manousos
• [Asterisk-Users] Overhead Paging   Paul Zimm
• [Asterisk-Users] Echo on polycom sip phone   Sean Garland
• [Asterisk-Users] <<< AGI and EXEC function CONFIRMATION >>>   Clayton Smith
• [Asterisk-Users] Astricon hotel recommendations.....?   Nick Barnes
• [Asterisk-Users] Asterisk+IVR functions trouble   Michael Manousos
• [Asterisk-Users] need help with zaptel.   Johannes van Hulst
• [Asterisk-Users] chan_oh323 and cdr   Roger Schreiter
• [Asterisk-Users] Echo on polycom sip phone   John Baker
• [Asterisk-Users] Asterisk+IVR functions trouble   Oleg A. Arkhangelsky
• [Asterisk-Users] ISDN Card Recommendation   Paterson, Mark
• [Asterisk-Users] <<< AGI and EXEC function CONFIRMATION >>>   jeff quade
• [Asterisk-Users] Error Compiling MySQL Friends   imail
• [Asterisk-Users] Astricon hotel recommendations.....?   Steven Sokol
• [Asterisk-Users] Echo on polycom sip phone   Chris HARIGA
• [Asterisk-Users] Anyone using Asterisk on Slackware 9?   Chris Blunt
• [Asterisk-Users] Sound card   Andrew Elchuk
• [Asterisk-Users] Sip Channel CLI   Alessio Focardi
• [Asterisk-Users] ISDN Card Recommendation   Rob Fugina
• [Asterisk-Users] Error Compiling MySQL Friends   R Wong
• [Asterisk-Users] Anyone using Asterisk on Slackware 9?   Andrew Kohlsmith
• [Asterisk-Users] Overhead Paging   Chris Shaw
• [Asterisk-Users] IPDialog call transfer   Jonathan
• [Asterisk-Users] Overhead Paging   Joseph
• [Asterisk-Users] ISDN Card Recommendation   Roger Schreiter
• [Asterisk-Users] RC2 and VoicePulse   Michael Welter
• [Asterisk-Users] Overhead Paging   Deon Rodden
• [Asterisk-Users] Advice on BT ISDN Services (UK)   Nick Barnes
• [Asterisk-Users] RC2 and Netmeeting 3.01 ?   Robert Rozman
• [Asterisk-Users] RC2 and VoicePulse   Marcelo Pacheco
• [Asterisk-Users] ShoreTel (ShoreLine) branded phones work with Asterisk?   Joseph Finley
• [Asterisk-Users] ISDN Card Recommendation   Lyle Giese
• [Asterisk-Users] RC2 and Netmeeting 3.01 ?   Michael Manousos
• [Asterisk-Users] No signal from ISDN-phone connected to hfc card in NT-mode   oliver_mlisten at gmx.de
• [Asterisk-Users] Overhead Paging   Tor Roberts
• [Asterisk-Users] Overhead Paging   John Bohman
• [Asterisk-Users] Overhead Paging   Tor Roberts
• [Asterisk-Users] Overhead Paging   Jay Milk
• [Asterisk-Users] ISDN Card Recommendation   Justin Huff
• [Asterisk-Users] Advice on BT ISDN Services (UK)   Linus Surguy
• [Asterisk-Users] Linux keeps deleting the ZAP files??   Shawn Parker
• [Asterisk-Users] Linux keeps deleting the ZAP files??   Steven Critchfield
• [Asterisk-Users] Zaptel/Zapata and SIP relationship   Lyle Giese
• [Asterisk-Users] ISDN Card Recommendation   Rob Fugina
• [Asterisk-Users] YAAN (Yet Another Asterisk Newbie)   Lyle Giese
• [Asterisk-Users] ISDN Card Recommendation   Justin Huff
• [Asterisk-Users] ISDN Card Recommendation   Rob Fugina
• [Asterisk-Users] Re: Compressing a dialplan   Maron Kristófersson
• [Asterisk-Users] home lab setup   Kevin Walsh
• [Asterisk-Users] Newbie needs help - Dev_Kit_Lite installation problem   David Luong
• [Asterisk-Users] Overhead Paging   Brian Pavane
• [Asterisk-Users] TDM400P lockups (FXO)   Michael George
• [Asterisk-Users] Error in app_callingcard.c with database connection   DIPAK PAUL
• [Asterisk-Users] No signal from ISDN-phone connected to hfc card in NT-mode   Oliver Brandt
• [Asterisk-Users] Asterisk mysql database   Jeremy Kenney
• [Asterisk-Users] Overhead Paging   Chris Shaw
• [Asterisk-Users] Overhead Paging   Chris Shaw
• [Asterisk-Users] PLC (Packet loss cancel) questions   Jorge Verastegui G
• [Asterisk-Users] Newbie needs help - Dev_Kit_Lite installationproblem   Dan Long
• [Asterisk-Users] g729 codec   voiprm
• [Asterisk-Users] Asterisk mysql database   Carlos Chavez
• [Asterisk-Users] ISDN Card Recommendation   Kimble Young
• [Asterisk-Users] Error Compiling MySQL Friends   imail
• [Asterisk-Users] ISDN Card Recommendation   Kimble Young
• [Asterisk-Users] Closing bug reports without fixing the repor ted problem   Rob Wise
• [Asterisk-Users] sip to sip calls thru asterisk   Lyle Giese
• [Asterisk-Users] TDM400P Problems   Greg Hulands
• [Asterisk-Users] Polycom non-IP phones   el Flynn
• [Asterisk-Users] TDM400P Problems   Lyle Giese
• [Asterisk-Users] Polycom SoundPoint IP 300 w/ SIP Software - Where Can I Get One?   Matthew Marlowe
• [Asterisk-Users] RC2 and Netmeeting 3.01 ?   Robert Rozman
• [Asterisk-Users] TDM400P lockups (FXO)   Michael George
• [Asterisk-Users] TDM400P lockups (FXO)   Marty Mastera
• [Asterisk-Users] TDM400P Problems   Greg Hulands
• [Asterisk-Users] PLC (Packet loss cancel) questions   Kevin Walsh
• [Asterisk-Users] Polycom non-IP phones   Kevin Walsh
• [Asterisk-Users] Polycom non-IP phones   el Flynn
• [Asterisk-Users] Problems with ISDN (NT-Mode) - Error Messages inside   Marco Czudej
• [Asterisk-Users] Number and name for SIP extension at the same time ?   Kevin Walsh
• [Asterisk-Users] Polycom non-IP phones   Kevin Walsh
• [Asterisk-Users] TDM400P Problems   Lyle Giese
• [Asterisk-Users] PLC (Packet loss cancel) questions   matt.riddell at sineapps.com
• [Asterisk-Users] Telemarketer screening   Mark Woods
• [Asterisk-Users] * and VoIP for LD   Chris Johnson
• [Asterisk-Users] TDM400P Problems   Greg Hulands
• [Asterisk-Users] TDM400P Problems   Lyle Giese
• [Asterisk-Users] VoIP Telephony with Asterisk book   Joseph Shi
• [Asterisk-Users] Asterisk mysql database   Jeremy Kenney
• [Asterisk-Users] Echo on polycom sip phone   John Baker
• [Asterisk-Users] TDM400P Problems   Adam Goryachev
• [Asterisk-Users] Telemarketer screening   William Suffill
• [Asterisk-Users] Telemarketer screening   Mark Woods
• [Asterisk-Users] TDM400P Problems   Greg Hulands
• [Asterisk-Users] Newbie with IAX2   paul at freestylenetworks.com
• [Asterisk-Users] Asterisk to Vonage   Ask Bjørn Hansen
• [Asterisk-Users] Hey admin: Do we have to have a 92-char reply-to header?   Brian Capouch
• [Asterisk-Users] PLC (Packet loss cancel) questions   Kevin Walsh
• [Asterisk-Users] PLC (Packet loss cancel) questions   matt.riddell at sineapps.com
• [Asterisk-Users] Re: Echo SIP-T100P-PRI   Ryan Thrash
• X-SBClass: Spam ( Re: [Asterisk-Users] TDM400P Problems)   PHP Mechanic
• [Asterisk-Users] 1stwave   Thomas Kuepper
• [Asterisk-Users] Re: Asterisk WITH Swyx... Any Idea?   Roberto Piola
• [Asterisk-Users] system reboot often?   Richard Scobie
• [Asterisk-Users] Advice on BT ISDN Services (UK)   Nick Barnes
• [Asterisk-Users] g729 codec   Andreas Sikkema
• [Asterisk-Users] Digit detect during a Background() inside a Macro wrongly jumps b ack to the calling context to match digits?   Kris Boutilier
• [Asterisk-Users] Digit detect during a Background() inside a Macro wrongly jumps b ack to the calling context to match digits?   el Flynn
• [Asterisk-Users] Hangup() doesn't always when talking to Nortel Norstar over CT1 E &M wink-start trunk line?   Kris Boutilier
• [Asterisk-Users] Advice on BT ISDN Services (UK)   Jon Fautley
• [Asterisk-Users] Digit detect during a Background() inside a Macro wrongly jumps b ack to the calling context to match digits?   Kris Boutilier
• [Asterisk-Users] Re: Asterisk mysql database   Daniel Niasoff
• [Asterisk-Users] 'set verbose 3' or other way to get '-vvv' level debugging out of running background asterisk?   Kris Boutilier
• [Asterisk-Users] Updated app_mysql.c, enabling use of INSERT and UPDATE   Andreas Sikkema
• [Asterisk-Users] Can't flash 7960: P0S30200 .bin not found   Tracy R Reed
• [Asterisk-Users] PLC (Packet loss cancel) questions   steve at daviesfam.org
• [Asterisk-Users] FXO interfaces used in UK?   David Gurr
• [Asterisk-Users] Sound card   Jason Williams
• [Asterisk-Users] Sip Channel CLI   Jason Williams
• [Asterisk-Users] Re: Can't flash 7960: P0S30200 .bin not found   Pavel Jezek
• [Asterisk-Users] Sip Channel CLI   Alessio Focardi
• [Asterisk-Users] Advice on BT ISDN Services (UK)   Jason Williams
• [Asterisk-Users] FXO interfaces used in UK?   Jason Williams
• [Asterisk-Users] Advice on BT ISDN Services (UK)   Benjamin Johnson
• [Asterisk-Users] PLC (Packet loss cancel) questions   Michael Manousos
• [Asterisk-Users] Error Compiling MySQL Friends   Bob Goddard
• [Asterisk-Users] PLC (Packet loss cancel) questions   Kevin Walsh
• [Asterisk-Users] TDM400P lockups (FXO)   Michael George
• [Asterisk-Users] Touch tone problem   Hall, Eric M.
• [Asterisk-Users] PLC (Packet loss cancel) questions   Michael Manousos
• [Asterisk-Users] PLC (Packet loss cancel) questions   steve at daviesfam.org
• [Asterisk-Users] PLC (Packet loss cancel) questions   Kevin Walsh
• [Asterisk-Users] Need help to install ISDN Fritz card   Radu Padure
• [Asterisk-Users] Using regular expression in dialplan   Selim
• [Asterisk-Users] Asterisk to Vonage   Deon Rodden
• [Asterisk-Users] Re: Echo SIP-T100P-PRI   Deon Rodden
• [Asterisk-Users] Queues - CallbackLoging Automaically?   Andrew Brown
• [Asterisk-Users] Cisco 7940 - SCCP or SIP?   slwatts at winckworths.co.uk
• [Asterisk-Users] Asterisk to Vonage   Doug Shubert
• [Asterisk-Users] Re: Can't flash 7960: P0S30200 .bin not found   Deon Rodden
• OT re: [Asterisk-Users] sip change?   Matt Schulte
• [Asterisk-Users] Asterisk to Vonage   Deon Rodden
• [Asterisk-Users] Using regular expression in dialplan   Selim
• [Asterisk-Users] sip change?   Deon Rodden
• [Asterisk-Users] Asterisk to Vonage   Doug Shubert
• [Asterisk-Users] Using regular expression in dialplan   Selim
• [Asterisk-Users] Asterisk compatible E1 cards   Vikram Rangnekar
• [Asterisk-Users] Voicetronix Segmentation Fault   Lex Lethol
• [Asterisk-Users] h323 with Fedora 2 & GCC 3.3   Marcin Mazurek
• [Asterisk-Users] Asterisk compatible E1 cards   Marcelo Pacheco
• [Asterisk-Users] how to fetch a call?   Roger Schreiter
• [Asterisk-Users] sip change?   Rich Adamson
• [Asterisk-Users] Problems dialing out with T100P and Adtran   Shawn Parker
• [Asterisk-Users] Re: how to fetch a call?   Tony Mountifield
• [Asterisk-Users] Cisco 7940 - SCCP or SIP?   Lex Lethol
• [Asterisk-Users] Re: Asterisk compatible E1 cards   Vikram Rangnekar
• [Asterisk-Users] Sound card   Andrew Elchuk
• [Asterisk-Users] how to fetch a call?   Rob Fugina
• [Asterisk-Users] Re: Asterisk compatible E1 cards   Marcelo Pacheco
• [Asterisk-Users] Cisco 7940 Sip Firmware   Benjamin Johnson
• [Asterisk-Users] Re: sip change? (Rich Adamson)   Walter Klomp
• [Asterisk-Users] Asterisk to Vonage   Jay Milk
• [Asterisk-Users] ACD ringall + roundrobin   Christopher L. Wade
• [Asterisk-Users] sip change?   Rich Adamson
• [Asterisk-Users] Queue Announcement not until after # accept call pressed   Andrew Brown
• [Asterisk-Users] Overhead Paging   Ejay Hire
• [Asterisk-Users] xlite Problems   Tim Jackson
• [Asterisk-Users] auto-gain, or different gain between incoming and outgoing calls (EURO ISDN PRI) ?   Walter Klomp
• [Asterisk-Users] Cisco 7940 SIP Firmware - Help.   Benjamin Johnson
• [Asterisk-Users] Broadvoice User hung up on voicemail   Kevin
• [Asterisk-Users] Problem dialing out to Free World Dialup   Andrew Elchuk
• [Asterisk-Users] Asterisk to Vonage   Chris Shaw
• [Asterisk-Users] Release 1.01 of FWD Assistant available (bugfix release)   Sunrise Ltd
• [Asterisk-Users] Problem dialing out to Free World Dialup   steve at daviesfam.org
• [Asterisk-Users] Broadvoice User hung up on voicemail   Chris Shaw
• [Asterisk-Users] questions and recommendations   Mark Phillips
• [Asterisk-Users] Overhead Paging   Chris Shaw
• [Asterisk-Users] IAX2 --> IAX2 confusion, it doesn't work...   Michael George
• [Asterisk-Users] FXOs   mgraves at mstvp.com
• [Asterisk-Users] Are there any graphic designers on this list?   Sunrise Ltd
• [Asterisk-Users] Newbie needs help - Dev_Kit_Lite installationproblem   David Luong
• [Asterisk-Users] Can a Macro call another Macro ?   Gary G. Hendershot
• [Asterisk-Users] GRSecurity and ALSA on a Gentoo Server   Ulexus
• [Asterisk-Users] Are there any graphic designers on this list?   Chris Shaw
• [Asterisk-Users] Problems dialing out with T100P and Adtran   Shawn Parker
• [Asterisk-Users] Re: sip change? (Rich Adamson)   Rich Adamson
• [Asterisk-Users] FXOs   slwatts at winckworths.co.uk
• [Asterisk-Users] No audio on PRI channel answered by Playback() or MeetMe()   Larry Shields
• [Asterisk-Users] Hey admin: Do we have to have a 92-char reply-to header?   Steve Szmidt
• [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe()   Robert Jackson
• [Asterisk-Users] FXOs   Tom Neville
• [Asterisk-Users] Someone please try MeetMe MOH with latest CVS and GS phone   Tony Mountifield
• [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe()   Scott Stingel
• [Asterisk-Users] Asterisk & Max TNTs   Ken Wiesner
• [Asterisk-Users] Advice on BT ISDN Services (UK)   Linus Surguy
• [Asterisk-Users] Advice on BT ISDN Services (UK)   Linus Surguy
• [Asterisk-Users] Overhead Paging   Rich Adamson
• [Asterisk-Users] Help with a fax via Grandstream Handytone 286?   Shawn Parker
• [Asterisk-Users] Re: how to fetch a call? (Tony Mountifield)   Sudhir Kumar
• [Asterisk-Users] FXOs   Rich Adamson
• [Asterisk-Users] Asterisk Assistants Custom Icon   Sunrise Ltd
• [Asterisk-Users] libr2   Vikram Rangnekar
• [Asterisk-Users] Speech Recognition and Asterisk   Mike Meyer
• [Asterisk-Users] Overhead Paging   Jay Milk
• [Asterisk-Users] sip change?   Jerry Roy
• [Asterisk-Users] FXOs   Marcelo Mercio Dandrea
• [Asterisk-Users] sip change?   Chad Brown
• [Asterisk-Users] sip change?   Doug Shubert
• [Asterisk-Users] Faxing with SPANDSP or any other mean ? Is itpossible ? Am I dreaming ?   Jean-François Rousseau
• [Asterisk-Users] Faxing with SPANDSP or any other mean ? Is itpossible ? Am I dreaming ?   Jean-François Rousseau
• [Asterisk-Users] Faxing with SPANDSP or any other mean ? Is itpossible ? Am I dreaming ?   Jean-François Rousseau
• [Asterisk-Users] Speech Recognition and Asterisk   Joe Phillips
• [Asterisk-Users] Faxing with SPANDSP or any other mean ? Isitpossible ? Am I dreaming ?   Chris Shaw
• [Asterisk-Users] Faxing with SPANDSP or any other mean ?Isitpossible ? Am I dreaming ?   Jean-François Rousseau
• [Asterisk-Users] AGI dtmf problems (with x-lite)   Raul Elizondo (wizardteam)
• [Asterisk-Users] Can a Macro call another Macro ?   Kevin Walsh
• [Asterisk-Users] regex and gotoif question   arno
• [Asterisk-Users] Overhead Paging   Rich Adamson
• [Asterisk-Users] xlite Problems   Jason Brockman
• [Asterisk-Users] iaxtel and jitterbuffer   Michael George
• [Asterisk-Users] Retrieve Info from Cisco Call Manager   Asterisk
• [Asterisk-Users] Speech Recognition and Asterisk   mattf
• [Asterisk-Users] Can a Macro call another Macro ?   Jay Milk
• X-SBClass: Spam ( Re: [Asterisk-Users] TDM400P Problems)   Lyle Giese
• [Asterisk-Users] Broadvoice User hung up on voicemail   Michael Swan
• [Asterisk-Users] Cisco 7940 SIP   Christopher L. Wade
• [Asterisk-Users] IAX dialing indication tone   spkao
• [Asterisk-Users] Cisco 7940 SIP   Brian Pavane
• [Asterisk-Users] * and VoIP for LD   Kanuri, Seshu
• [Asterisk-Users] iaxtel and jitterbuffer   Kris Boutilier
• [Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()   Larry Shields
• [Asterisk-Users] IAX2 --> IAX2 confusion, it doesn't work...   Michael George
• [Asterisk-Users] Re: Retrieve Info from Cisco Call Manager   Pavel Jezek
• [Asterisk-Users] Re: Someone please try MeetMe MOH with latest CVS and GS phone   Tony Mountifield
• [Asterisk-Users] does agi wait for digit work in a meetme room ?   Eric Bart
• [Asterisk-Users] IAXy Power in Australia?   Jeremy Bogan
• [Asterisk-Users] AGI dtmf problems (with x-lite)   Steven Critchfield
• [Asterisk-Users] Can a Macro call another Macro ?   Kevin Walsh
• [Asterisk-Users] Speech Recognition and Asterisk   Steven Critchfield
• [Asterisk-Users] IAXy Power in Australia?   Steven Critchfield
• [Asterisk-Users] IAXy Power in Australia?   Sean Cheesman
• [Asterisk-Users] iaxtel and jitterbuffer   Michael George
• [Asterisk-Users] mysql-vm-routines and Directory app   Carlos Chavez
• [Asterisk-Users] Zap & ANSWER the Call   Rodrigo P. Telles
• [Asterisk-Users] FXO interfaces used in UK?   George Gardiner
• [Asterisk-Users] "7" Dialing gives a busy signal   Don Moskaluk
• [Asterisk-Users] "7" Dialing gives a busy signal   Lyle Giese
• [Asterisk-Users] "7" Dialing gives a busy signal   Don Moskaluk
• [Asterisk-Users] "7" Dialing gives a busy signal   Don Moskaluk
• [Asterisk-Users] Digit detect during a Background() inside a Macro wrongly jumps b ack to the calling context to match digits?   Kris Boutilier
• [Asterisk-Users] "7" Dialing gives a busy signal   Lyle Giese
• [Asterisk-Users] "7" Dialing gives a busy signal   Don Moskaluk
• [Asterisk-Users] how does one get the very best quality output?   Clayton Smith
• [Asterisk-Users] "7" Dialing gives a busy signal   Don Moskaluk
• [Asterisk-Users] "7" Dialing gives a busy signal   Lyle Giese
• [Asterisk-Users] "7" Dialing gives a busy signal   Lyle Giese
• [Asterisk-Users] IAXy Power in Australia?   Gary
• [Asterisk-Users] Disconnection From IAXTel   Muiz Motani
• [Asterisk-Users] Disconnection From IAXTel   Muiz Motani
• [Asterisk-Users] how does one get the very best quality output?   Darren Wiebe
• [Asterisk-Users] FXOs   Jamie Carl
• [Asterisk-Users] Someone please try MeetMe MOH with latest CVS and GS phone   Dave Cotton
• [Asterisk-Users] iaxtel and jitterbuffer   Brian McSpadden
• [Asterisk-Users] IAXy Power in Australia?   Jeremy Bogan
• [Asterisk-Users] FXO probs in Aus. Should I give up?   Jamie Carl
• [Asterisk-Users] IAXy Power in Australia?   Gary
• [Asterisk-Users] ISDN BRI card exepriences in UK   David Gurr
• [Asterisk-Users] IAXy Power in Australia?   Florian Overkamp
• [Asterisk-Users] IAXy Power in Australia?   Roy Eddleston
• [Asterisk-Users] IAXy Power in Australia?   Jeremy Bogan
• [Asterisk-Users] IAXy Power in Australia?   Roy Eddleston
• [Asterisk-Users] G729 licenses   Sergey Lapin
• [Asterisk-Users] Disconnection From IAXTel   Sunrise Ltd
• [Asterisk-Users] chan_sccp2 & 7960 -- documentation and examplerequest.   Craig Guy
• [Asterisk-Users] AGI dtmf problems (with x-lite) (solved)   Raul Elizondo (wizardteam)
• [Asterisk-Users] IAXy Power in Australia?   Jeremy Bogan
• [Asterisk-Users] incomming call rejected using IAX2 with FWD   Storm D. J. Petersen
• [Asterisk-Users] incomming call rejected using IAX2 with FWD   Storm D. J. Petersen
• [Asterisk-Users] BT Communicator (SIP???) and Asterisk   gARetH baBB
• [Asterisk-Users] Queue Announcement not until after # accept call pressed   Tim Robinson
• [Asterisk-Users] Queue Announcement not until after # accept call pressed   Tim Robinson
• [Asterisk-Users] switch statement in extensions.conf   Michael George
• [Asterisk-Users] IAXy Power in Australia?   Duane
• [Asterisk-Users] Re: Disconnection from IAXTel   Brad Ediger
• [Asterisk-Users] iaxtel and jitterbuffer   Rich Adamson
• [Asterisk-Users] G729 licenses   Kevin Walsh
• [Asterisk-Users] iaxtel and jitterbuffer   Scott Laird
• [Asterisk-Users] incomming call rejected using IAX2 with FWD   steveb at asbc.demon.co.uk
• [Asterisk-Users] incomming call rejected using IAX2 with FWD   Lyle Giese
• [Asterisk-Users] IAX dialing indication tone (PI = 8)   Matthew Oulton
• [Asterisk-Users] incomming call rejected using IAX2 with FWD   Bill Seddon
• [Asterisk-Users] asterisks and vonage   Michael Di Martino
• [Asterisk-Users] FXO interfaces used in UK?   Rich Adamson
• [Asterisk-Users] Queue Announcement not until after # accept callpressed   Edward Eastman
• [Asterisk-Users] UK Disconnect supervision with TDM400P   Edward Eastman
• [Asterisk-Users] SIP Provider for Reseller   Beierlein Moritz
• [Asterisk-Users] incomming call rejected using IAX2 with FWD   Sunrise Ltd
• [Asterisk-Users] Queue Announcement not until after # accept call pressed   Greg Boehnlein
• [Asterisk-Users] Broadvoice problem   Russell Horn
• [Asterisk-Users] Broadvoice problem   Marty Mastera
• [Asterisk-Users] Broadvoice problem   Ed Brady
• [Asterisk-Users] Broadvoice problem   Ed Brady
• [Asterisk-Users] Broadvoice problem   Marty Mastera
• [Asterisk-Users] Broadvoice problem   Marty Mastera
• [Asterisk-Users] Broadvoice problem   Ed Brady
• [Asterisk-Users] Broadvoice problem   Ed Brady
• [Asterisk-Users] POE   Steve Maroney
• [Asterisk-Users] POE   Michael Welter
• [Asterisk-Users] iaxtel and jitterbuffer   Michael George
• [Asterisk-Users] iaxtel and jitterbuffer   Michael George
• [Asterisk-Users] iaxtel and jitterbuffer   Michael Graves
• [Asterisk-Users] POE   Greg Boehnlein
• [Asterisk-Users] Broadvoice BYOD Plans   Ben Wern
• [Asterisk-Users] UK Disconnect supervision with TDM400P   Richard Scobie
• [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting   Ben Wern
• [Asterisk-Users] POE   Michael Welter
• [Asterisk-Users] Disconnection From IAXTel   Muiz Motani
• [Asterisk-Users] iaxtel and jitterbuffer   Andrew Kohlsmith
• [Asterisk-Users] IAXy Power in Australia?   Jeremy Bogan
• [Asterisk-Users] iaxtel and jitterbuffer   Andrew Kohlsmith
• [Asterisk-Users] POE   Greg Boehnlein
• [Asterisk-Users] SIP Provider for Reseller   Roger Schreiter
• [Asterisk-Users] SIP Provider for Reseller   sgup015 at ec.auckland.ac.nz
• [Asterisk-Users] Newbie   Michael Di Martino
• [Asterisk-Users] Newbie   Michael Graves
• [Asterisk-Users] Newbie   Greg Broiles
• [Asterisk-Users] POE   Craig Guy
• [Asterisk-Users] SIP Provider for Reseller   Chad Brown
• [Asterisk-Users] asterisks and vonage   Paterson, Mark
• [Asterisk-Users] POE   Kevin Day
• [Asterisk-Users] Are there any graphic designers on this list?   Paterson, Mark
• [Asterisk-Users] POE   Steven Pritchard
• [Asterisk-Users] Distinctive ring detection problem   Paul Budden
• [Asterisk-Users] iaxtel and jitterbuffer   Michael George
• [Asterisk-Users] POE   Kevin P. Fleming
• [Asterisk-Users] iaxtel and jitterbuffer   Andrew Kohlsmith
• [Asterisk-Users] IAX dialing indication tone (PI = 8)   spkao
• [Asterisk-Users] Disconnection From IAXTel   Dave Cotton
• [Asterisk-Users] iaxtel and jitterbuffer   steve at daviesfam.org
• [Asterisk-Users] iaxtel and jitterbuffer   steve at daviesfam.org
• [Asterisk-Users] where can I find spandsp?   Rich Adamson
• [Asterisk-Users] Disconnection From IAXTel   Muiz Motani
• [Asterisk-Users] Disconnection From IAXTel   Dave Cotton
• [Asterisk-Users] Disconnection From IAXTel   Adam Hart
• [Asterisk-Users] Mobile phone integration via bluetooth   hg at grasser-edv.de
• [Asterisk-Users] IAXy Power in Australia?   Roy Eddleston
• [Asterisk-Users] Newbie   Steve Totaro
• [Asterisk-Users] Termination in Holland.   micke at party.pp.se
• [Asterisk-Users] Disconnection From IAXTel   Muiz Motani
• [Asterisk-Users] IAXy died   Wilson Pickett
• [Asterisk-Users] Asterisk Assistants for Linux or Windoze???   Sunrise Ltd
• [Asterisk-Users] Disconnection From IAXTel   Sunrise Ltd
• [Asterisk-Users] Re: where can I find spandsp?   Stefan Tichy
• [Asterisk-Users] IAXy Power in Australia?   Duane
• [Asterisk-Users] Caller ID problem   M Wahid Ullah
• [Asterisk-Users] Extens and number converting so that i can dial following one standaard.   Johannes van Hulst
• [Asterisk-Users] Termination in Holland.   Johannes van Hulst
• [Asterisk-Users] iaxtel and jitterbuffer   Andrew Kohlsmith
• [Asterisk-Users] iaxtel and jitterbuffer   Andrew Kohlsmith
• [Asterisk-Users] Mobile phone integration via bluetooth   Scott Laird
• [Asterisk-Users] iaxtel and jitterbuffer   Linus Surguy
• [Asterisk-Users] iaxtel and jitterbuffer   joachim
• [Asterisk-Users] Re: Distinctive ring detection problem   David Cook
• [Asterisk-Users] iaxtel and jitterbuffer   Michael George
• [Asterisk-Users] iaxtel and jitterbuffer   Linus Surguy
• [Asterisk-Users] iaxtel and jitterbuffer   Michael George
• [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn   Sergio Serrano
• [Asterisk-Users] PLC (Packet loss cancel) questions   Steve Underwood
• [Asterisk-Users] Empty Queues   Ben Merrills
• [Asterisk-Users] VoIP Telephony with Asterisk book   Steve Underwood
• [Asterisk-Users] how does one get the very best quality output?   Nicholas Bachmann
• [Asterisk-Users] PLC (Packet loss cancel) questions   matt.riddell at sineapps.com
• [Asterisk-Users] Sip device not login or register calls to that device go to busy voicemail not un-available   Ariel's Hotmail
• [Asterisk-Users] Sip device not login or register calls to that device go to busy voicemail not un-available   Maxim Litnitsky
• [Asterisk-Users] Servers   Steve Maroney
• [Asterisk-Users] Servers   Duane Cox
• [Asterisk-Users] Servers   Andrew Kohlsmith
• [Asterisk-Users] Sip device not login or register calls to that device go to busy voicemail not un-available   Rich Adamson
• [Asterisk-Users] iaxtel and jitterbuffer   steve at daviesfam.org
• [Asterisk-Users] iaxtel and jitterbuffer   steve at daviesfam.org
• [Asterisk-Users] iaxtel and jitterbuffer   Andrew Kohlsmith
• [Asterisk-Users] Jitter buffer   steve at daviesfam.org
• [Asterisk-Users] Python and AGI   Tracy R Reed
• [Asterisk-Users] iaxtel and jitterbuffer   steve at daviesfam.org
• [Asterisk-Users] iaxtel and jitterbuffer   Michael George
• [Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()   Larry Shields
• [Asterisk-Users] not getting ringing/busy/answer feedback on my PRI   Larry Shields
• [Asterisk-Users] IAXy Power in Australia?   Roy Eddleston
• [Asterisk-Users] not getting ringing/busy/answer feedback on my PRI   Larry Shields
• [Asterisk-Users] Jitter buffer   Andrew Kohlsmith
• [Asterisk-Users] SMS and asterisk   Maxim Litnitsky
• [Asterisk-Users] Servers   Steve Maroney
• [Asterisk-Users] System freezes when using Festival with usecache   Ed Brady
• [Asterisk-Users] Revert to dial tone?   Greg Blakely
• [Asterisk-Users] Which Zaptel release goes with Asterisk-1.0-RC2 ???   Sunrise Ltd
• [Asterisk-Users] SMS & Asterisk   Maxim Litnitsky
• [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting   James Jones
• [Asterisk-Users] Broadvoice problem   James Jones
• [Asterisk-Users] iaxtel and jitterbuffer   Kris Boutilier
• [Asterisk-Users] Asterisk H.323 channel...   Hello World
• [Asterisk-Users] Static Problem (t100p - Channel Bank)   Brent Franks
• [Asterisk-Users] Sip device not login or register calls to that device go to busy voicemail not un-available   Craig Guy
• [Asterisk-Users] Bridging audio in cmd_dial() before connect completes?   Kris Boutilier
• [Asterisk-Users] Asterisk and codecs?   Balgansuren Batsukh
• [Asterisk-Users] POE   asteriskstuff at ziplip.com
• [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting   Kevin
• [Asterisk-Users] Zap channels stuck in 'Rsrvd' state   Shane Young
• [Asterisk-Users] VoIP Telephony with Asterisk book   Joseph Shi
• [Asterisk-Users] Re: Distinctive ring detection problem   Paul Budden
• [Asterisk-Users] PLC (Packet loss cancel) questions   Steve Underwood
• [Asterisk-Users] ${CONTEXT}   Steve Maroney
• [Asterisk-Users] Still unacceptable echo on X101P   David Cook
• [Asterisk-Users] PLC (Packet loss cancel) questions   matt.riddell at sineapps.com
• [Asterisk-Users] Queue Announcement not until after # acceptcallpressed   csm-lists at csma.biz
• [Asterisk-Users] AgentCallbackLogin by other means   csm-lists at csma.biz
• [Asterisk-Users] zaptel configuration   Imran Akbar
• [Asterisk-Users] Extens and number converting so that i can dial following one standaard.   Greg Hill
• [Asterisk-Users] Help debugging voicemail problem   Lex Lethol
• [Asterisk-Users] zaptel configuration   Steve Maroney
• [Asterisk-Users] Mix Data and SIP Phones   yves at swissinfo.org
• [Asterisk-Users] ${CONTEXT}   Steven Critchfield
• [Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()   Tobias Jönsson
• [Asterisk-Users] Bridging audio in cmd_dial() before connect completes?   Peter Svensson
• [Asterisk-Users] Revert to dial tone?   Peter Svensson
• [Asterisk-Users] just-added second X100P   spectro
• [Asterisk-Users] G729 licenses   Andreas Sikkema
• [Asterisk-Users] X100 and call duration   Dan
• [Asterisk-Users] Jitter buffer   steve at daviesfam.org
• [Asterisk-Users] Voicetronix OpenLine4 immediately hangs up on every call   Sunrise Ltd
• [Asterisk-Users] Problem with modified-Prepaid-Application   DIPAK PAUL
• [Asterisk-Users] How can i convert number with exten   Johannes van Hulst
• [Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()   Steven Critchfield
• [Asterisk-Users] How can i convert number with exten   Tim Robinson
• [Asterisk-Users] Mix Data and SIP Phones   Steve Totaro
• [Asterisk-Users] FXOs   Steve Totaro
• [Asterisk-Users] How can i convert number with exten   Steve Totaro
• [Asterisk-Users] G729 licenses   Brian Wilkins
• [Asterisk-Users] IAX.conf problem (NEWBIE ALERT!)   Roy Eddleston
• [Asterisk-Users] SIP and IAX Registration Problem with Dynamic IP   Axel Eble
• [Asterisk-Users] G729 licenses   Brian Wilkins
• [Asterisk-Users] SMS & Asterisk   Axel Eble
• [Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()   Tobias Jönsson
• [Asterisk-Users] AVM Fritz on Fedora 1   Radu Padure
• [Asterisk-Users] Suitable for Dynamic IVR Platform?   Paul Dugas
• [Asterisk-Users] zaphfc success stories   Tobias Jönsson
• [Asterisk-Users] Re: zaptel configuration   Atif Rasheed
• [Asterisk-Users] VoIP Telephony with Asterisk book   Leif Madsen
• [Asterisk-Users] Still unacceptable echo on X101P   Rich Adamson
• [Asterisk-Users] Re: Distinctive ring detection problem   David Cook
• [Asterisk-Users] Still unacceptable echo on X101P   Duane
• [Asterisk-Users] AgentCallbackLogin by other means   Nicolas Gudino
• [Asterisk-Users] Mix Data and SIP Phones   Jon Bebeau
• [Asterisk-Users] New Error Log Messages   Brent Franks
• [Asterisk-Users] FXOs   Rich Adamson
• [Asterisk-Users] Broadvoice disconnecting on Asterisk Voice Mail   Kevin
• [Asterisk-Users] Re: Distinctive ring detection problem   Paul Budden
• [Asterisk-Users] libr2   Steve Underwood
• [Asterisk-Users] PLC (Packet loss cancel) questions   Kevin Walsh
• [Asterisk-Users] chan_oh323 loading error   Jonas Lindvall (AllTele)
• [Asterisk-Users] Suitable for Dynamic IVR Platform?   Umar Sear
• [Asterisk-Users] Still unacceptable echo on X101P   Michael Graves
• [Asterisk-Users] G729 licenses   Kevin Walsh
• [Asterisk-Users] G729 licenses   Andreas Sikkema
• [Asterisk-Users] chan_oh323 loading error   Oleg A. Arkhangelsky
• [Asterisk-Users] Voiceronix and asterisk   Kelvin And Lisa
• [Asterisk-Users] G729 licenses   Andrew Kohlsmith
• [Asterisk-Users] Parsing problem with oh323   Jonas Lindvall (AllTele)
• [Asterisk-Users] Follow me   Vladyslav
• [Asterisk-Users] Still unacceptable echo on X101P   Kevin Walsh
• [Asterisk-Users] H323 outgoing calls   Darren Wiebe
• [Asterisk-Users] How does call routing actually work with SIP?   Daryll Strauss
• [Asterisk-Users] G729 licenses   Kevin Walsh
• [Asterisk-Users] Still unacceptable echo on X101P   Duane
• [Asterisk-Users] G729 licenses   Kevin Walsh
• [Asterisk-Users] Broadvoice disconnecting on Asterisk Voice Mail   Chris Shaw
• [Asterisk-Users] G729 licenses   Andreas Sikkema
• [Asterisk-Users] chan_oh323 loading error   Michael Manousos
• [Asterisk-Users] Sip device not login or register calls tothat device go to busy voicemail not un-available   Chris Shaw
• [Asterisk-Users] Still unacceptable echo on X101P   Rich Adamson
• [Asterisk-Users] chan_oh323 loading error   Roger Schreiter
• [Asterisk-Users] G729 licenses   Marcelo Pacheco
• [Asterisk-Users] Still unacceptable echo on X101P   Kevin Walsh
• [Asterisk-Users] IAX.conf problem (NEWBIE ALERT!)   Bill Seddon
• [Asterisk-Users] Cisco 7940 SIP   Chris A. Icide
• [Asterisk-Users] How does call routing actually work with SIP?   Kevin Walsh
• [Asterisk-Users] X100 and call duration   Lyle Giese
• [Asterisk-Users] G729 licenses   Stefan de Konink
• [Asterisk-Users] Suitable for Dynamic IVR Platform?   Bill Seddon
• [Asterisk-Users] How can i convert number with exten   Kevin Walsh
• [Asterisk-Users] Broadvoice disconnecting on Asterisk Voice Mail   Greg Hill
• [Asterisk-Users] Does anyone have a working GR-303 config?   Chris Jensen
• [Asterisk-Users] Problems with T100P card not releasing channels.   Shawn Parker
• [Asterisk-Users] Still unacceptable echo on X101P   Mike Benoit
• [Asterisk-Users] Revert to dial tone?   Greg Blakely
• [Asterisk-Users] Polycom SoundPoint IP 300 Configuration   Matthew Marlowe
• [Asterisk-Users] G729 licenses   Brian Capouch
• [Asterisk-Users] Problems with T100P card not releasing channels.   Scott Stingel
• [Asterisk-Users] Polycom SoundPoint IP 300 Configuration   mattf
• [Asterisk-Users] Broadvoice disconnecting on Asterisk Voice Mail   Chris Shaw
• [Asterisk-Users] G729 licenses   Andrew Kohlsmith
• [Asterisk-Users] Anyone used Tecom IP2004 SIP phone?   David Gurr
• [Asterisk-Users] Polycom SoundPoint IP 300 Configuration   Matthew Marlowe
• [Asterisk-Users] Cisco 7940 SIP   Christopher L. Wade
• [Asterisk-Users] Faxing with an IAXy   Ben Merrills
• [Asterisk-Users] Revert to dial tone?   Peter Svensson
• [Asterisk-Users] SIPJack   Muiz Motani
• [Asterisk-Users] Polycom SoundPoint IP 300 Configuration   mattf
• [Asterisk-Users] Redirect SIP calls to the SIP provider sipgate.de   Johannes van Hulst
• [Asterisk-Users] Snom Programmable button Mini Howto and ring state patch   David Hinkle
• [Asterisk-Users] How record conversation to sound file ?   lst.ara
• [Asterisk-Users] How does call routing actually work with SIP?   Daryll Strauss
• [Asterisk-Users] Still unacceptable echo on X101P   Brent Franks
• [Asterisk-Users] Problems with T100P card not releasing channels.   Shawn Parker
• [Asterisk-Users] zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0)   Vasyl Rublyov
• [Asterisk-Users] G729 licenses   Eric Wieling
• [Asterisk-Users] Polycom SoundPoint IP 300 Configuration   Eric Wieling
• [Asterisk-Users] SOLVED - Problems compiling zaptel driver   PBX Portela
• [Asterisk-Users] Polycom SoundPoint IP 300 Configuration   Matthew Marlowe
• [Asterisk-Users] MWI Light On SoundPoint IP 300   Matthew Marlowe
• [Asterisk-Users] Polycom SoundPoint IP 300 Configuration   Matthew Marlowe
• [Asterisk-Users] PLC (Packet loss cancel) questions   Steve Kann
• [Asterisk-Users] How does call routing actually work with SIP?   Kevin Walsh
• [Asterisk-Users] Asterisk with Sayson 480 ADSI   Craig Neumanns
• [Asterisk-Users] PLC (Packet loss cancel) questions   Chris Shaw
• [Asterisk-Users] number of simultaneous calls with E&M   muhumuza brian
• [Asterisk-Users] number of simultaneous calls with E&M   Michael Welter
• [Asterisk-Users] Still unacceptable echo on X101P   Michael Graves
• [Asterisk-Users] FXOs   Michael Graves
• [Asterisk-Users] Still unacceptable echo on X101P   Kevin Walsh
• [Asterisk-Users] Snom Programmable button Mini Howto and ring state patch   John Todd
• [Asterisk-Users] Revert to dial tone?   Greg Blakely
• [Asterisk-Users] G729 licenses   Umar Sear
• [Asterisk-Users] AstriCon Reminder: Please register today   Steven Sokol
• [Asterisk-Users] FXOs   Ron Frederick
• [Asterisk-Users] Compile error H323   Enrico Stahn
• [Asterisk-Users] Re: does agi wait for digit work in a meetme room ?   Eric Bart
• [Asterisk-Users] [Fwd: [Asterisk-Dev] Snom Programmable button Mini Howto and ring state patch]   David Hinkle
• [Asterisk-Users] PLC (Packet loss cancel) questions   Chris Shaw
• [Asterisk-Users] PLC (Packet loss cancel) questions   Chris Shaw
• [Asterisk-Users] SIPJack   Muiz Motani
• [Asterisk-Users] Delays while playing a message   Alex Zarubin
• [Asterisk-Users] Polycom SoundPoint... Gains - Which is for speakerphone   Matthew Marlowe
• [Asterisk-Users] number of simultaneous calls with E&M   Kris Boutilier
• [Asterisk-Users] Re: New to Asterisk and a question   Brad Stockdale
• [Asterisk-Users] Re: New to Asterisk and a question   Tim Jackson
• [Asterisk-Users] Reload crashes Asterisk ?   Walter Klomp
• [Asterisk-Users] VoIP Telephony with Asterisk book   Lex Lethol
• [Asterisk-Users] Voiceronix and asterisk   Lex Lethol
• [Asterisk-Users] Voicetronix OpenLine4 immediately hangs up on every call   Lex Lethol
• [Asterisk-Users] Polycom SoundPoint... Gains - Which is for speakerphone   John Baker
• [Asterisk-Users] Asterisk and Citrix   Learning, Bill
• [Asterisk-Users] Asterisk and Citrix   Craig Guy
• [Asterisk-Users] VoicePulse Connect DTMF with IAX2   Bryce Nesbitt (mailing list account)
• [Asterisk-Users] VoicePulse Connect DTMF with IAX2   Brian Capouch
• [Asterisk-Users] My Three-way calls work backwards   Steve Maroney
• [Asterisk-Users] Newbie - Voicemail Password Help   Java Rockx
• [Asterisk-Users] How record conversation to sound file ?   Martin Holler
• AW: [Asterisk-Users] SMS & Asterisk   Michael Labuschke
• [Asterisk-Users] transferring call to another line   Imran Akbar
• [Asterisk-Users] Cisco 7940 SIP   slwatts at winckworths.co.uk
• [Asterisk-Users] Do not get calldeflection (capiCD) to work.   Nicolas
• [Asterisk-Users] How record conversation to sound file ?   Radek Terber
• AW: [Asterisk-Users] SMS & Asterisk   Axel Eble
• [Asterisk-Users] Newbie - Voicemail Password Help   Stephen Hon
• [Asterisk-Users] Jitter over Sat   Storm D. J. Petersen
• [Asterisk-Users] Transfer to queue   Ben Merrills
• [Asterisk-Users] Jitter over Sat   Martin Mielke
• [Asterisk-Users] Polycom SoundPoint... Gains - Which isfor speakerphone   Matthew Marlowe
• [Asterisk-Users] ** ASTRICON * LAST CALL FOR REGISTRATION   Olle E. Johansson
• [Asterisk-Users] Jitter over Sat   Storm D. J. Petersen
• [Asterisk-Users] which distro for asterisk?   Brian Wilkins
• [Asterisk-Users] Transfer from MOH to MOH doesn't work.   Michael Løjtnant
• [Asterisk-Users] Compile error H323   Michael Manousos
• [Asterisk-Users] pattern matching problems   Florian Overkamp
• [Asterisk-Users] pattern matching problems   Atif Rasheed
• [Asterisk-Users] pattern matching problems   Adam Goryachev
• [Asterisk-Users] Newbie - Voicemail Password Help   Lyle Giese
• [Asterisk-Users] Jitter over Sat   Bob Goddard
• [Asterisk-Users] extensions => s,1,Dial(Zap/2/number) noise   Imran Akbar
• [Asterisk-Users] Jitter over Sat   Storm D. J. Petersen
• [Asterisk-Users] PLC (Packet loss cancel) questions   Steve Underwood
• [Asterisk-Users] Re: pattern matching problems   Atif Rasheed
• [Asterisk-Users] BRI numbers   Altus Snyman
• AW: AW: [Asterisk-Users] SMS & Asterisk   Michael Labuschke
• AW: AW: [Asterisk-Users] SMS & Asterisk   Axel Eble
• AW: AW: AW: [Asterisk-Users] SMS & Asterisk   Michael Labuschke
• [Asterisk-Users] SIP registration with public dynamic ip address   tonini.massimo at tiscali.it
• [Asterisk-Users] limit the length of extensions   Deon Rodden
• [Asterisk-Users] which distro for asterisk?   Tzafrir Cohen
• [Asterisk-Users] Polycom SoundPoint... Gains - Which is for speakerphone   Michael Graves
• [Asterisk-Users] Harddisk noise on TE410P   Claus Futtrup
• [Asterisk-Users] VoicePulse Connect DTMF with IAX2   Deon Rodden
• [Asterisk-Users] which distro for asterisk?   Deon Rodden
• [Asterisk-Users] Harddisk noise on TE410P   Deon Rodden
• [Asterisk-Users] incomming call rejected using IAX2 with FWD   Deon Rodden
• [Asterisk-Users] newbie question about PBX Call Pickup   Maurizio Marini
• [Asterisk-Users] Re: zaptel configuration   Imran Akbar
• [Asterisk-Users] which distro for asterisk?   Tim Jackson
• [Asterisk-Users] VoicePulse Connect DTMF with IAX2   Michael Welter
• [Asterisk-Users] which distro for asterisk?   Huddleston, Robert
• [Asterisk-Users] X100P Questions: Voicemail and Phone Port questions   Matt G
• [Asterisk-Users] Re: Newbie - Voicemail Password Help   Jason Kawakami
• [Asterisk-Users] Polycom IP 300 - Displaying Only Caller NAME... What about NUMBER?   Matthew Marlowe
• [Asterisk-Users] Polycom SoundPoint... Gains - Whichis for speakerphone   Matthew Marlowe
• [Asterisk-Users] which distro for asterisk?   Tzafrir Cohen
• [Asterisk-Users] Polycom SoundPoint... Gains - Which is for speakerphone   John Baker
• [Asterisk-Users] Polycom SoundPoint... Gains -Which is for speakerphone   Matthew Marlowe
• [Asterisk-Users] Polycom IP 300 - Displaying Only Caller NAME... What about NUMBER?   Eric Wieling
• [Asterisk-Users] Can i send calling costs to a SIP IP phone display   Johannes van Hulst
• [Asterisk-Users] answer from wrong port   Benjamin Lawetz
• [Asterisk-Users] Polycom IP 300 - Displaying Only CallerNAME... What about NUMBER?   Matthew Marlowe
• [Asterisk-Users] Polycom IP 300 - Displaying Only CallerNAME... What about NUMBER?   Eric Wieling
• [Asterisk-Users] Harddisk noise on TE410P   Scott Stingel
• [Asterisk-Users] PLC (Packet loss cancel) questions   Chris Shaw
• [Asterisk-Users] VoicePulse Connect DTMF with IAX2   Marty Mastera
• [Asterisk-users] PLC (Packet loss cancel) questions   Chris Shaw
• [Asterisk-Users] PSTN noob question   Nick W
• [Asterisk-Users] VoicePulse Connect DTMF with IAX2   Arkadi Shishlov
• [Asterisk-Users] limit the length of extensions   Lyle Giese
• [Asterisk-Users] VoicePulse Connect DTMF with IAX2   Michael Welter
• [Asterisk-Users] limit the length of extensions   Deon Rodden
• [Asterisk-Users] Harddisk noise on TE410P   Claus Futtrup
• [Asterisk-Users] Harddisk noise on TE410P   Scott Stingel
• [Asterisk-Users] Re: Newbie - Voicemail Password Help   Java Rockx
• [Asterisk-Users] limit the length of extensions   Lyle Giese
• [Asterisk-Users] limit the length of extensions   Chris Shaw
• [Asterisk-Users] multiple lines with SIP like MGCP?   Matthew Boehm
• [Asterisk-Users] PSTN noob question   Rich Adamson
• [Asterisk-Users] Snom Programmable button Mini Howto and ringstate patch   David Hinkle
• [Asterisk-Users] multiple lines with SIP like MGCP?   Chris Shaw
• [Asterisk-Users] Losing voice on Digium demo server - how to spot problem ?   Robert Rozman
• [Asterisk-Users] Analog lines and TDM card   Marcello Lupo
• [Asterisk-Users] error: CDR on channel '<unknown>' has not started   eduardo at garnier.inf.br
• [Asterisk-Users] detect telco voicemail stutter-tone   Ryan Courtnage
• [Asterisk-Users] Can asterisk detect BUSY signal?   eduardo at garnier.inf.br
• [Asterisk-Users] IAX Client   Jon Bebeau
• [Asterisk-Users] limit the length of extensions   Deon Rodden
• [Asterisk-Users] Can asterisk detect BUSY signal?   Andrew Thompson
• [Asterisk-Users] Can asterisk detect BUSY signal?   mattf
• [Asterisk-Users] multiple lines with SIP like MGCP?   Rich Adamson
• [Asterisk-Users] limit the length of extensions   Adam Goryachev
• [Asterisk-Users] Going to voicemail instead of queue if no agent is logged in ?   Robert Rozman
• [Asterisk-Users] OT: Headset for Cisco 7960?   Nate Carlson
• [Asterisk-Users] OT: Headset for Cisco 7960?   Bryan Vyhmeister
• [Asterisk-Users] Streaming an audio file to a Zap channel before answer   Tim Robinson
• [Asterisk-Users] OT: Headset for Cisco 7960?   Benjamin Johnson
• [Asterisk-Users] OT: Headset for Cisco 7960?   Edward Eastman
• [Asterisk-Users] T100P No D-channels   Pliva, Josef
• [Asterisk-Users] OT: Headset for Cisco 7960?   Dan Austin
• [Asterisk-Users] OT: Headset for Cisco 7960?   James H. Thompson
• [Asterisk-Users] T100P No D-channels   Chris Shaw
• [Asterisk-Users] OT: Headset for Cisco 7960?   B. J. Bomar
• [Asterisk-Users] T100P No D-channels   Pliva, Josef
• [Asterisk-Users] Why is it called 'Comedian Mail?   Kris Boutilier
• [Asterisk-Users] Hardware suggestion   Manfred Petz
• [Asterisk-Users] T100P No D-channels   Chris Shaw
• [Asterisk-Users] OT: Headset for Cisco 7960?   Nate Carlson
• [Asterisk-Users] T100P No D-channels   Eric Wieling
• [Asterisk-Users] Why is it called 'Comedian Mail?   Kevin Walsh
• [Asterisk-Users] Why is it called 'Comedian Mail?   Chris Shaw
• [Asterisk-Users] Can asterisk detect BUSY signal?   Kevin Walsh
• [Asterisk-Users] OT: Headset for Cisco 7960?   Rich Adamson
• [Asterisk-Users] Why is it called 'Comedian Mail?   TC
• [Asterisk-Users] IAX Client   Michael Van Donselaar
• [Asterisk-Users] Why is it called 'Comedian Mail?   Kevin Walsh
• [Asterisk-Users] Can asterisk detect BUSY signal?   Andrew Kohlsmith
• [Asterisk-Users] Can asterisk detect BUSY signal?   Chris Shaw
• [Asterisk-Users] Why is it called 'Comedian Mail?   Philip Fleischer
• [Asterisk-Users] Newbie - Voicemail Password Help   Umar Sear
• [Asterisk-Users] Newbie - Voicemail Password Help   Bob Goddard
• [Asterisk-Users] Zap & ANSWER the Call   Rodrigo P. Telles
• [Asterisk-Users] Asterisk SIP between two networks   Sergio Serrano
• [Asterisk-Users] Newbie - Voicemail Password Help   Stephen Hon
• [Asterisk-Users] Can only call asterisk once   James Doherty
• [Asterisk-Users] good Dutch TTS ?   Danny Zak
• [Asterisk-Users] Asterisk codecs and packet size   Luis Vazquez
• [Asterisk-Users] Zap & ANSWER the Call   Lyle Giese
• [Asterisk-Users] Cisco 79XX SIP Ring Tones   Christopher L. Wade
• [Asterisk-Users] MP3Player strange error   Maxim Litnitsky
• [Asterisk-Users] Asterisk codecs and packet size   Kevin Walsh
• AW: AW: [Asterisk-Users] SMS & Asterisk   Maxim Litnitsky
• [Asterisk-Users] SMS & Asterisk - an explanation   Scott Stingel
• [Asterisk-Users] PLC (Packet loss cancel) questions   Steve Kann
• [Asterisk-Users] SIP registration with public dynamic ip address   Lyle Giese
• [Asterisk-Users] Polycom SoundPoint IP 300 Configuration   Reid A. Forrest
• [Asterisk-Users] Losing voice on Digium demo server - how to spotproblem ?   Steve Totaro
• [Asterisk-Users] Analog lines and TDM card   Steve Totaro
• [Asterisk-Users] T100P Configuration for Mixed Voice & Data   Shawn Kelley
• [Asterisk-Users] All you polycom folks.....   Brent Franks
• [Asterisk-Users] Polycom SoundPoint IP 300 Configuration   Brent Franks
• [Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()   Kevin P. Fleming
• [Asterisk-Users] install software version to mediatrix 1204 (how to)   eder tan
• [Asterisk-Users] Does anyone have a working GR-303 config?   Kevin P. Fleming
• [Asterisk-Users] OT: Headset for Cisco 7960?   Shaun Ewing
• [Asterisk-Users] Line death not recognized on TDM400P?   matt.riddell at sineapps.com
• [Asterisk-Users] Dial/Zap doesn't work   Imran Akbar
• [Asterisk-Users] Line death not recognized on TDM400P?   matt.riddell at sineapps.com
• [Asterisk-Users] iaxtel and jitterbuffer   steve at daviesfam.org
• [Asterisk-Users] Jitter over Sat   steve at daviesfam.org
• [Asterisk-Users] DeadAGI Application   Darren Wiebe