August 2006 Archives by date

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Starting: Tue Aug 1 00:02:23 MST 2006
Ending: Thu Aug 31 23:46:24 MST 2006
Messages: 3082

• [asterisk-users] Re: question about asterisk DB   Tzafrir Cohen
• [asterisk-users] VoipNow 1.2.0 Beta   Michiel van Baak
• [asterisk-users] cmd DIAL - Who picked up the call?   Kai Ober
• [asterisk-users] SRTP help   Khaled Chehab
• [asterisk-users] Permission for files generated by voicemail   Jean-Yves Avenard
• [asterisk-users] asterisk gui   vivek at staff.ownmail.com
• [asterisk-users] asterisk gui   Rajeev Natarajan
• [asterisk-users] SoftHangup with Polycom_acd_functions release of asterisk   Dean at INKnBITs
• [asterisk-users] asterisk gui   Alex Robar
• [asterisk-users] MWI from Asterisk to Meridian   Andrew Kohlsmith
• [asterisk-users] Re: Re: FYI - first release of alarm response code.   Steven
• [asterisk-users] nat and qualify questions   BerkHolz, Steven
• [asterisk-users] asterisk gui   Rajeev Natarajan
• [asterisk-users] asterisk gui   Alex Robar
• [asterisk-users] MWI from Asterisk to Meridian   Andrew Kohlsmith
• [asterisk-users] Is there a smarter way to ban expensive calls in dial plan?   Chris Blunt
• [asterisk-users] Is there a smarter way to ban expensive calls in dial plan?   Marco Mouta
• [asterisk-users] AddQueueMember and Local channel   Asterisk
• [asterisk-users] Is there a smarter way to ban expensive calls indial plan?   Martin Schrott - Thinking-Systems
• [asterisk-users] IAX and Accountcode   Joshua Colp
• [asterisk-users] Missing Fast AGI calling 'h' exten without hanging up   Tony Mountifield
• [asterisk-users] MWI from Asterisk to Meridian   Johann Steinwendtner
• [asterisk-users] Help debugging strange asterisk behaviour   jan.sarin at securia.se
• [asterisk-users] Missing Fast AGI calling 'h' exten without hanging up   Rich Adamson
• [asterisk-users] Media direct from IAX Phone to IAX Phone   Kamran Ahmad
• [asterisk-users] SV: Help debugging strange asterisk behaviour   jan.sarin at securia.se
• [asterisk-users] nat and qualify questions   Marco Mouta
• [asterisk-users] Problem with distortion of initial voicemail prompt   Frank Tarczynski
• [asterisk-users] Re: Missing Fast AGI calling 'h' exten without hanging up   Tony Mountifield
• [asterisk-users] Park / ParkAndAnnounce   Guillermo Roditi
• [asterisk-users] SendText() & displaying text messages onaSIPhandset's screen   Guillermo Roditi
• [asterisk-users] IAX and Accountcode   Joshua Colp
• [asterisk-users] Multi Asterisk Server to relay call request   Stephen Wingfield
• [asterisk-users] can't retake call after dialing through Zap/E1 wich doesn't answer   Manrique Feoli
• [asterisk-users] Extend analog phone via SIP (OT)   Ira
• [asterisk-users] IAX over two T1 connections bad quality   Tim Panton
• [asterisk-users] Controllable hold music   Thomas Kenyon
• [asterisk-users] nat and qualify questions   Alyed Tzompa
• [asterisk-users] Codec selection / IAX tunnels   Thomas Kenyon
• [asterisk-users] cmd DIAL - Who picked up the call?   Koopmann, Jan-Peter
• [asterisk-users] IAX and Accountcode   Douglas Garstang
• [asterisk-users] MWI from Asterisk to Meridian   kritikus Araklidas
• [asterisk-users] IAX and Accountcode   Thomas Kenyon
• [asterisk-users] MWI from Asterisk to Meridian   Johann Steinwendtner
• [asterisk-users] Extend analog phone via SIP (OT)   Rich Adamson
• [asterisk-users] Re: How to configure NOKIA N70 with Asterisk?   Benny Amorsen
• [asterisk-users] MySQL 5.0+ and the MySQL addon - Can use stored procedures?   Rushowr
• [asterisk-users] Line drops   J. Oquendo
• [asterisk-users] Re: Operator Console(s)/Shared Call Appearances   Mr. Jones
• [asterisk-users] IAX and Accountcode   Douglas Garstang
• [asterisk-users] AddQueueMember and Local channel   Asterisk
• [asterisk-users] ISDN incoming call - inband info and announcements BEFORE ANSWER   Michal Dole ž el
• [asterisk-users] rx_fax problem   Paradise Dove
• [asterisk-users] Is there a smarter way to ban expensive calls in dial plan?   Andrea Spadaccini
• [asterisk-users] Dundi and Dial Arguments   Mitch Sharp
• [asterisk-users] IAX and Accountcode   Tim Panton
• [asterisk-users] IAX and Accountcode   Douglas Garstang
• [asterisk-users] Help debugging strange asterisk behaviour   Mojo with Horan & Company, LLC
• [asterisk-users] Re: How to configure NOKIA N70 with Asterisk?   FaberK
• [asterisk-users] Unicall stack, right versions?   Barzilai
• [asterisk-users] Dundi and Dial Arguments   Douglas Garstang
• [asterisk-users] Dundi and Dial Arguments   Michiel van Baak
• [asterisk-users] codec conversion   Wasif
• [asterisk-users] Dundi and Dial Arguments   Douglas Garstang
• [asterisk-users] Polycom IP600 HTTP Provisioning problem   VaibhaV Sharma
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Pablo Mora
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   C F
• [asterisk-users] codec conversion   Russell Bryant
• [asterisk-users] FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found   Zen Kato
• [asterisk-users] Asterisk with VoIP phone   J Rangi
• [asterisk-users] ISDN incoming call - inband info and announcements BEFORE ANSWER   Matthew Fredrickson
• [asterisk-users] VOIP phone for Receptionist use   Jeff Busch
• [asterisk-users] RE: asterisk-users Digest, Vol 25, Issue 2   AstATN
• [asterisk-users] VOIP phone for Receptionist use   Bill Gibbs
• [asterisk-users] ANNOUNCE: libss7   Matthew Fredrickson
• [asterisk-users] Unicall stack, right versions?   Moises Silva
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Jorge Mendoza
• [asterisk-users] VOIP phone for Receptionist use   DM
• [asterisk-users] Unicall stack, right versions?   Steve Underwood
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Pablo Mora
• [asterisk-users] MWI from Asterisk to Meridian   C F
• [asterisk-users] A2Billing - destination   Luciano Moreira
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   C F
• [asterisk-users] Re: AEL2 Looping   Steve Murphy
• [asterisk-users] Unicall stack, right versions?   Barzilai Spinak
• [asterisk-users] softhangup() problem   Shaun Hofer
• [asterisk-users] cmd DIAL - Who picked up the call?   Eric "ManxPower" Wieling
• [asterisk-users] SER local as an Asterisk Trunk   Nhadie Ramos
• [asterisk-users] Re: [Serusers] SER local as an Asterisk Trunk   ram
• [asterisk-users] Dundi and Dial Arguments   Michiel van Baak
• SV: [asterisk-users] VOIP phone for Receptionist use   Jon Schøpzinsky
• [asterisk-users] Problem with Cisco7970 SIP load / call transfer   Juha Suhonen
• [asterisk-users] Asterisk config with Analouge Audio Codec model number MP108FXS   Mr shobhit nirala
• [asterisk-users] SRTP   harrygaillac-sip at yahoo.fr
• [asterisk-users] Slow dialing from PBX via E1   Gavin Hamill
• SV: [asterisk-users] Help debugging strange asterisk behaviour   jan.sarin at securia.se
• [asterisk-users] Asterisk config with Analouge Audio Codec model number MP108FXS   ram
• [asterisk-users] rx_fax problem   Steve Hanselman
• SV: [asterisk-users] Help debugging strange asterisk behaviour   jan.sarin at securia.se
• [asterisk-users] Asterisk with VoIP phone   Bruce Reeves
• [asterisk-users] newbie - suggestions on installing Asterisk for SOHO   Pele Zico
• [asterisk-users] Unicall stack, right versions?   Steve Underwood
• [asterisk-users] Follow ON calling on DISA   sdcharly at gmail.com
• [asterisk-users] FXO module burn out !?   Rostislav Bagrov
• [asterisk-users] canreinvite=yes and RTP dropping in and out   Joshua Colp
• [asterisk-users] FXO module burn out !?   jacobso1
• [asterisk-users] FXO module burn out !?   Andrew Kohlsmith
• [asterisk-users] Unicall stack, right versions?   leonimar cape
• [asterisk-users] Limitations of IAX   Joshua Colp
• [asterisk-users] polycom soundstation 501 crash   Stas Khromoy
• [asterisk-users] canreinvite=yes and RTP dropping in and out   Joshua Colp
• [asterisk-users] Dell Poweredge 1950 / 2950   Frédéric Marti
• [asterisk-users] Call Routing based on Caller-Id   Matthew Crocker
• [asterisk-users] Playback() does not work   Camilo Echeverry
• [asterisk-users] polycom soundstation 501 crash   Alexander Lopez
• [asterisk-users] polycom soundstation 501 crash   Stas Khromoy
• [asterisk-users] Unicall stack, right versions?   Moises Silva
• [asterisk-users] polycom soundstation 501 crash   Jessee J Holmes
• [asterisk-users] polycom soundstation 501 crash   Stas Khromoy
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Pablo Mora
• [asterisk-users] Limitations of IAX   Joshua Colp
• [asterisk-users] FXO module burn out !?   asterisk
• [asterisk-users] polycom soundstation 501 crash   Sebastian Milioto
• [asterisk-users] polycom soundstation 501 crash   Jessee J Holmes
• [asterisk-users] Asterisk, Linksys SPA-3000 echo   Dean at INKnBITs
• [asterisk-users] Arrays ???   Pele Zico
• [asterisk-users] polycom soundstation 501 crash   Jim Rice
• [asterisk-users] Issue with IAX2 and Real Time configuration   Facundo Ameal
• [asterisk-users] [ANN] - Coder Needed for Patch   Bart Fisher
• [asterisk-users] Re: Issue with IAX2 and Real Time configuration   Facundo Ameal
• [asterisk-users] FXO module burn out !?   Alex Robar
• [asterisk-users] SIP_HEADER() read-only   Vincent Regnard
• [asterisk-users] Asterisk, Linksys SPA-3000 echo   Rich Adamson
• [asterisk-users] Arrays ???   Benchev
• [asterisk-users] Limitations of IAX   Douglas Garstang
• [asterisk-users] canreinvite=yes and RTP dropping in and out   Gary Richardson
• [asterisk-users] DTMF intermittent on menu.   Shane Burrell
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   C F
• [asterisk-users] asterisk optimizing   Jack Wei
• [asterisk-users] sip phone networking question [possibly OT]   Mojo with Horan & Company, LLC
• [asterisk-users] cmd DIAL - Who picked up the call?   Koopmann, Jan-Peter
• [asterisk-users] asterisk optimizing   tracinet
• [asterisk-users] sip phone networking question [possibly OT]   Colin Anderson
• [asterisk-users] canreinvite=yes and RTP dropping in and out   Gary Richardson
• [asterisk-users] Strange behavior with SIP registration/connectivity   Ronald Lewis
• [asterisk-users] cmd DIAL - Who picked up the call?   Vadim Berezniker
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Jorge Mendoza
• [asterisk-users] GSM analogue router   Garth van Sittert
• [asterisk-users] Rookie question, trying to learn   Randy Paries
• [asterisk-users] Re: Arrays ???   Pele Zico
• [asterisk-users] Limitations of IAX   Douglas Garstang
• [asterisk-users] Limitations of IAX   Andrew Kohlsmith
• [asterisk-users] IAX Trunking   Douglas Garstang
• [asterisk-users] cmd DIAL - Who picked up the call?   Koopmann, Jan-Peter
• [asterisk-users] Limitations of IAX   Douglas Garstang
• [asterisk-users] cmd DIAL - Who picked up the call?   Vadim Berezniker
• [asterisk-users] RemoveQueueMember isn't working.   Keith Herrington
• [asterisk-users] creidt card processing sripts for asterisk   Joseph
• [asterisk-users] Limitations of IAX   Andrew Kohlsmith
• [asterisk-users] creidt card processing sripts for asterisk   Jon Pounder
• [asterisk-users] DUNDi with SIP   Douglas Garstang
• [asterisk-users] creidt card processing sripts for asterisk   Joseph
• [asterisk-users] DUNDi with SIP   Aaron Daniel
• [asterisk-users] Asterisk with VoIP phone   shadowym
• [asterisk-users] VOIP phone for Receptionist use   shadowym
• [asterisk-users] creidt card processing sripts for asterisk   Jon Pounder
• [asterisk-users] DUNDi with SIP   Douglas Garstang
• [asterisk-users] creidt card processing sripts for asterisk   Mojo with Horan & Company, LLC
• [asterisk-users] VOIP phone for Receptionist use   Dr. Michael J. Chudobiak
• [asterisk-users] Re: DUNDi with SIP   Tony Mountifield
• [asterisk-users] DUNDi with SIP   Aaron Daniel
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] DUNDi with SIP   Douglas Garstang
• [asterisk-users] unsubscribe   Keith Herrington
• [asterisk-users] Ateus Easy gate call progress   Jan Fousek
• [asterisk-users] DUNDi with SIP   Alex Robar
• [asterisk-users] DUNDi with SIP   Douglas Garstang
• [asterisk-users] DUNDi with SIP   Douglas Garstang
• [asterisk-users] creidt card processing sripts for asterisk   Joseph
• [asterisk-users] DUNDi with SIP   Alex Robar
• [asterisk-users] DUNDi with SIP   Douglas Garstang
• [asterisk-users] chan_zap.c: Failed to read gains: Invalid argument   jan.sarin at securia.se
• [asterisk-users] Re: need a pointer regarding scripting asterisk   Andy Kuo
• [asterisk-users] RE: Asterisk with VoIP phone (shadowym)   J Rangi
• [asterisk-users] Re: DUNDi with SIP   Tony Mountifield
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] need dialout help in python script   shawn bright
• [asterisk-users] Re: DUNDi with SIP   Aaron Daniel
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Aaron Daniel
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Aaron Daniel
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Rookie question, trying to learn   Time Bandit
• [asterisk-users] Clocking Multiple T1 Cards   Kevin P. Fleming
• [asterisk-users] create custom cdr's   Kevin P. Fleming
• [asterisk-users] need dialout help in python script   Shidan
• [asterisk-users] Binary/unreadable configuration files?   Kevin P. Fleming
• [asterisk-users] Caller ID on Transfers   Kevin P. Fleming
• [asterisk-users] IP JitterBuffer for 1.2.5   Thierry Querette
• [asterisk-users] Unicall stack, right versions?   Steve Underwood
• [asterisk-users] Call Routing based on Caller-Id   Leo Ann Boon
• [asterisk-users] Call Routing based on Caller-Id   Brandon Galbraith
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] Polycom 501 : How to make it ring when already on a call   Mike
• Subject: [asterisk-users] Slow dialing from PBX via E1   AstATN
• [asterisk-users] Asterisk 1.0.1 in SuSE 10.0   Oscar Carrillo
• [asterisk-users] About Digium cards and HP DL servers   Angel Gomez
• [asterisk-users] Asterisk 1.0.1 in SuSE 10.0   Russell Bryant
• [asterisk-users] Polycom 501 : How to make it ring when already ona call   Brian Vincent (C)
• [Asterisk-Users] Duration for billing   Steve Edwards
• [asterisk-users] Number of Rings Before Asterisk Takes Over   Joe Pokupec
• [asterisk-users] Is there a smarter way to ban expensive calls in dial plan?   William Piper
• [asterisk-users] create custom cdr's   William Piper
• [asterisk-users] Number of Rings Before Asterisk Takes Over   Eric "ManxPower" Wieling
• [asterisk-users] [OT] FYI: Polycom phone intermittent disconnects   Jeff Turner
• [asterisk-users] Number of Rings Before Asterisk Takes Over   Paul Hales
• [asterisk-users] need dialout help in python script   shawn bright
• [asterisk-users] Re: need a pointer regarding scripting asterisk   Benjamin Stocker
• [asterisk-users] cmd DIAL - Who picked up the call?   Koopmann, Jan-Peter
• [asterisk-users] BRIDGEPEER and DIALEDPEERNAME empty   Koopmann, Jan-Peter
• [asterisk-users] Number of Rings Before Asterisk Takes Over   Joe Pokupec
• [asterisk-users] Number of Rings Before Asterisk Takes Over   Ira
• [asterisk-users] Rookie question, trying to learn   Randy Paries
• [asterisk-users] Number of Rings Before Asterisk Takes Over   Massimo Nuvoli
• [asterisk-users] IAX2 Trunking CPU usage   Jon Schøpzinsky
• [asterisk-users] Rookie question, trying to learn   Marco Mouta
• Subject: [asterisk-users] Slow dialing from PBX via E1   Gavin Hamill
• [asterisk-users] MWI from Asterisk to Meridian   Leo Ann Boon
• [asterisk-users] Garbled initial voicemail prompt   Frank Tarczynski
• [asterisk-users] queue in realtime   unplug
• [asterisk-users] SIP_HEADER() read-only   Vincent Regnard
• [asterisk-users] How to check if channel varaible have been set/not empty?   Jan du Toit
• [asterisk-users] What I can use with ASTERISK to call clients to remind them about their appointments   dmitri smirnoff
• [asterisk-users] queue in realtime   Benjamin Stocker
• [asterisk-users] What I can use with ASTERISK to call clients to remind them about their appointments   Fabian Müller
• [asterisk-users] How to check if channel varaible have been set/notempty?   Rushowr
• [asterisk-users] SIP_HEADER() read-only   Joshua Colp
• [asterisk-users] MoH native volume   Joshua Colp
• [asterisk-users] wip 300 opensource code - changes to support SIPMESSAGE   Joshua Colp
• [asterisk-users] Ringing all extensions   Joshua Colp
• [asterisk-users] MWI from Asterisk to Meridian   kritikus Araklidas
• [asterisk-users] RAM memory high Comsumption   Joshua Colp
• [asterisk-users] Garbled initial voicemail prompt   Joshua Colp
• [asterisk-users] IAX2 Trunking CPU usage   Tim Panton
• [asterisk-users] Clocking Multiple T1 Cards   Andrew Kohlsmith
• [asterisk-users] How to check if channel varaible have been set/notempty?   Andrew Kohlsmith
• [asterisk-users] IAX Variables   Joshua Colp
• [asterisk-users] volume adjustment?   Kohler, Jeffrey
• [asterisk-users] [OT] FYI: Polycom phone intermittent disconnects   Bill Gibbs
• [asterisk-users] RAM memory high Comsumption   kritikus Araklidas
• [asterisk-users] Asterisk suddenly die   kritikus Araklidas
• [asterisk-users] Asterisk suddenly die   yusuf
• [asterisk-users] Ringing all extensions   J. Oquendo
• [asterisk-users] Polycom 501 : How to make it ring when alreadyona call   Mike
• [asterisk-users] Polycom 501 : How to make it ring when alreadyona call   Jerry Jones
• [asterisk-users] wip 300 opensource code - changes to support SIP MESSAGE   Jerry Geis
• [asterisk-users] Polycom 501 : How to make it ring whenalreadyona call   Mike
• [asterisk-users] MoH native volume   Carlos Chavez
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Asterisk H323 and Alcatel 4400   Carlos Chavez
• [asterisk-users] SIP_HEADER() read-only   Douglas Garstang
• [asterisk-users] Reboot Mediatrix   Julian Varanini
• [asterisk-users] IAX Trunking   Douglas Garstang
• [asterisk-users] Re: Arrays ???   Benchev
• [asterisk-users] VoiceMail being cutoff when leaving message   Randy Paries
• [asterisk-users] IAX Variables   Douglas Garstang
• [asterisk-users] Rookie question, trying to learn   Randy Paries
• [asterisk-users] SIP_HEADER() read-only   Eric "ManxPower" Wieling
• [asterisk-users] queue in realtime   unplug
• [asterisk-users] About Digium cards and HP DL servers   Anthony Rodgers
• [asterisk-users] [OT] FYI: Polycom phone intermittent disconnects   Anthony Rodgers
• [asterisk-users] IAX Variables   Douglas Garstang
• [asterisk-users] SIP_HEADER() read-only   Douglas Garstang
• [asterisk-users] Forbidden - wrong password on authentication for INVITE   Brent Torrenga
• [asterisk-users] Queue bug: When 2 callers call in, only one is processed until the first is answered   Keith Herrington
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] How to forward a call to an outside line   Dan Casey
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Detecting voicemail from CO on FXO port and passing to H.323 phone. Possible?   Bob Bosiljevac
• [asterisk-users] Detecting voicemail from CO on FXO port andpassing to H.323 phone. Possible?   Steven Totaro
• [asterisk-users] Re: How to forward a call to an outside line   Steven
• [asterisk-users] Detecting voicemail from CO on FXO port   Paul Davidson
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] About Digium cards and HP DL servers   angom at telnor.net
• [asterisk-users] About Digium cards and HP DL servers   VOICEIN at aol.com
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] Meetme echo in recording   Mike
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Using Flite in a call file.   Joey McDonald
• [asterisk-users] Run a script at certain CLI writes   Bart Fisher
• [asterisk-users] Run a script at certain CLI writes   Joey McDonald
• [asterisk-users] Detecting voicemail from CO on FXO port   C F
• [asterisk-users] Detecting voicemail from CO on FXO port andpassing to H.323 phone. Possible?   C F
• [asterisk-users] Echo cancell   Pablo Allietti
• [asterisk-users] Polycom 501 : How to make it ring whenalreadyona call   C F
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   C F
• [asterisk-users] Re: How to forward a call to an outside line   Dan Casey
• [asterisk-users] Run a script at certain CLI writes   Bart Fisher
• [asterisk-users] Re: How to forward a call to an outside line   Dan Casey
• [asterisk-users] Detecting voicemail from CO on FXO   Paul Davidson
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Pablo Mora
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Stephen Murphy
• [asterisk-users] Garbled initial voicemail prompt   Brian Capouch
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   C F
• [asterisk-users] trinary expression   John Williams
• [asterisk-users] Detecting voicemail from CO on FXO   C F
• [asterisk-users] Encoding recorded queue calls to mp3   jan.sarin at securia.se
• [asterisk-users] Problem dialing out with a TDB400P   Dante Passalacqua
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Pablo Mora
• [asterisk-users] New UK prompts   Steve Kennedy
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Jim Freeze
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Douglas Garstang
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   C F
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Avi Miller
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Brian Vincent (C)
• [asterisk-users] Run a script at certain CLI writes   Nick Hoffman
• [asterisk-users] Problem dialing out with a TDB400P   T. Shaw
• [asterisk-users] Run a script at certain CLI writes   Bruce Reeves
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Stephen Murphy
• [asterisk-users] MWI from Asterisk to Meridian   AstATN
• [asterisk-users] Re: How to forward a call to an outside line   Greg Broiles
• [asterisk-users] Ringing all extensions   AstATN
• [asterisk-users] Opinions on Rhino PCI FXO cards   shadowym
• [asterisk-users] Re: How to forward a call to an outside line   Dan Casey
• [asterisk-users] Using Flite in a call file.   Russell Bryant
• [asterisk-users] cannot received calls in pstn line   Lito Lampitoc
• [asterisk-users] Run a script at certain CLI writes   Eric "ManxPower" Wieling
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Eric "ManxPower" Wieling
• [asterisk-users] Strange behaviour Panasonic KX-TD1232   Eric "ManxPower" Wieling
• [asterisk-users] Run a script at certain CLI writes   Eric "ManxPower" Wieling
• [asterisk-users] cannot received calls in pstn line   Crazy Boy
• [asterisk-users] any experiece with SPHINX   Abubakar A. Khaliq
• [asterisk-users] Asterisk@Home, call reporting and performance   Esteban Guana-Jarrin
• [asterisk-users] SIP_HEADER() read-only   Vincent Regnard
• [asterisk-users] asterisk dosenot compile   vivek at staff.ownmail.com
• [asterisk-users] Polycom IP600 HTTP Provisioning problem   Dovid Bender
• [asterisk-users] asterisk dosenot compile   Jan Fousek
• [asterisk-users] Asterisk with AVM B1 and HFC   Marco Dieckhoff
• [asterisk-users] ANI agi   Sharon Lim
• [asterisk-users] Asterisk with AVM B1 and HFC   Peer Oliver Schmidt
• [asterisk-users] Echo cancell   Koopmann, Jan-Peter
• [asterisk-users] Unicall stack, right versions?   leonimar cape
• [asterisk-users] Configuring meetme recording quality (8kHz to 32kHz or higher)   Jan du Toit
• [asterisk-users] How to connect Snom softphone from my home?   Crazy Boy
• SV: [asterisk-users] Help debugging strange asterisk behaviour   jan.sarin at securia.se
• [asterisk-users] Re: DUNDi with SIP   Watkins, Bradley
• [asterisk-users] Load balancing of IAX2   Kamran Ahmad
• [asterisk-users] creidt card processing sripts for asterisk   Dennis Nacino
• [asterisk-users] Load balancing of IAX2   Florian Overkamp
• [asterisk-users] asterisk dosenot compile   Joshua Colp
• [asterisk-users] speech gaps with iax2   Pavel Jezek
• [asterisk-users] How to connect Snom softphone from my home?   Guido Hecken
• [asterisk-users] Re: Configuring meetme recording quality (8kHz to 32kHz or higher)   Tony Mountifield
• SV: [asterisk-users] Help debugging strange asterisk behaviour (update)   jan.sarin at securia.se
• [Asterisk-Users] How to configure NOKIA N70 with Asterisk?   Crazy Boy
• [asterisk-users] Re: Re: How to forward a call to an outside line   Steven
• [asterisk-users] Re: Run a script at certain CLI writes   Steven
• [asterisk-users] Re: Using Flite in a call file.   Steven
• [asterisk-users] Prevent a Polycom contact list to be overwritten   Jerry Jones
• [asterisk-users] sendtext() to another machine   Jerry Geis
• [asterisk-users] Asterisk, Linksys SPA-3000 echo   Neil Cherry
• [asterisk-users] speech gaps with iax2   Pavel Jezek
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Re: DUNDi with SIP   Steve Totaro
• [asterisk-users] About Digium cards and HP DL servers   Matt Riddell (NZ)
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Paul Zimm
• [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Asterisk Manager Vb 6   Dovid Bender
• [asterisk-users] SIP/Qualify   Pavel Jezek
• [asterisk-users] Sangoma A200 and Disconnected Cables   Andres
• [asterisk-users] SNOM 360   Dovid Bender
• [asterisk-users] SNOM 360   Dovid Bender
• [asterisk-users] Re: Re: How to forward a call to an outside line   Dan Casey
• [asterisk-users] SIP/Qualify   Rushowr
• [asterisk-users] SIP and podcasts   Dovid Bender
• [asterisk-users] SNOM 360   Steve Davies
• [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] How to forward a call to an outside line   Dovid Bender
• [asterisk-users] Using Flite in a call file.   Joey McDonald
• [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] Sangoma A200 and Disconnected Cables   Alex Robar
• [asterisk-users] SNOM 360   Steve Davies
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   Steve Totaro
• [asterisk-users] SIP/Qualify   Peder at NetworkOblivion
• [asterisk-users] AgentCallBackLogin+Queue   Gleidson Antonio Henriques
• [asterisk-users] Re: DUNDi with SIP   Douglas Garstang
• [asterisk-users] Jabber questions   Julian Lyndon-Smith
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   Jerry Jones
• [asterisk-users] AgentCallBackLogin+Queue   Guido Hecken
• [asterisk-users] Festival Not Working   Jon Scottorn
• [asterisk-users] Is the manager good for high traffic?? but only with one connection to it   Manrique Feoli
• [asterisk-users] Problems with monitor / mixmonitor stopping if using Local channels   Julian Lyndon-Smith
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   Steve Totaro
• [asterisk-users] SNOM 360   Julio Arruda
• [asterisk-users] Problems with monitor / mixmonitor stopping if using Local channels   Julian Lyndon-Smith
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   Jerry Jones
• [asterisk-users] Dialplan routing based on CallerID   Matthew Crocker
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   BJ Weschke
• [asterisk-users] Dialplan routing based on CallerID   C F
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] Steve Totaro I am trying to reach you.   Ferguson, Michael
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] Asterisk, Linksys SPA-3000 echo   Rich Adamson
• [asterisk-users] Is the manager good for high traffic?? but only with one connection to it   Matt Florell
• [asterisk-users] Running AGI in background   Bromont -
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   Steve Totaro
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] Running AGI in background   John Williams
• [asterisk-users] Mediatrix 1204 and Asterisk 1.2.10   Julian Varanini
• [asterisk-users] Steve Totaro I am trying to reach you.   Steve Totaro
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   BJ Weschke
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] DISA + Voicemail + DTMF   Dave
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   David Coulson
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] Asterisk and Siemens Legacy PBX   Jerry Jones
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] Simple config question   John Williams
• [asterisk-users] How to play music on hold from within PHP AGI scripts?   Leo Burd
• [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] Aastra VLAN issues   Kris Seraphine
• [asterisk-users] Setting CALLERID on a residential telco line   hugolivude
• [asterisk-users] Setting CALLERID on a residential telco line   Steven Ringwald
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Eric "ManxPower" Wieling
• [asterisk-users] Simple config question   Russell Bryant
• [asterisk-users] Setting CALLERID on a residential telco line   hugolivude
• [asterisk-users] Check call duration on active call in CLI?   voiplist
• [asterisk-users] Setting CALLERID on a residential telco line   VoIP Street
• [asterisk-users] SIP/Qualify   William Piper
• [asterisk-users] Setting CALLERID on a residential telco line   Eric "ManxPower" Wieling
• [asterisk-users] Detecting voicemail from CO on FXO port   Bob Bosiljevac
• [asterisk-users] Detecting voicemail from CO on FXO port   Rich Adamson
• [asterisk-users] Setting CALLERID on a residential telco line   Andrea Spadaccini
• [asterisk-users] autocreatepeer in iax   Kamran Ahmad
• [asterisk-users] autocreatepeer in iax   Russell Bryant
• [asterisk-users] Setting CALLERID on a residential telco line   Avi Miller
• [asterisk-users] Help - call recording being cut short if transferred   Julian Lyndon-Smith
• [asterisk-users] cisco 2600   FaberK
• [asterisk-users] Help - call recording being cut short if transferred   Julian J. M.
• [asterisk-users] MWI from Asterisk to Meridian   kritikus Araklidas
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] Help - call recording being cut short if transferred   Julian Lyndon-Smith
• [asterisk-users] [Solution] Call Asterisk from GoogleTalk and have it tell you the status of your IAX2 links.   Matt Riddell (NZ)
• [asterisk-users] IAXy behind NAT, unable to transfer?   Jeff Iddings
• [asterisk-users] Japanese Sound Files   Nhadie
• [asterisk-users] Setting CALLERID on a residential telco line   Eric "ManxPower" Wieling
• [asterisk-users] IAXy behind NAT, unable to transfer?   Joshua Colp
• [asterisk-users] Japanese Sound Files   Hermann Wecke
• [asterisk-users] IAXy behind NAT, unable to transfer?   Joshua Colp
• [asterisk-users] load average with MOH   Jon Farmer
• [asterisk-users] Setting CALLERID on a residential telco line   Hermann Wecke
• [asterisk-users] Check call duration on active call in CLI?   Hermann Wecke
• [asterisk-users] Check call duration on active call in CLI?   Jon Farmer
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] Asterisk and Siemens Legacy PBX   Jon Farmer
• [asterisk-users] IAXy behind NAT, unable to transfer?   Moises Silva
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] IAXy behind NAT, unable to transfer?   Jeff Iddings
• [asterisk-users] IAXy behind NAT, unable to transfer?   Jeff Iddings
• [asterisk-users] how to check the status of a channel   Thomas Artner
• [asterisk-users] how to check the status of a channel   Marcus Carlson
• [asterisk-users] how to check the status of a channel   Eric "ManxPower" Wieling
• [asterisk-users] Help with perl AGI script   Roy Kidder
• [asterisk-users] g729 and trafic   Walter Willis
• [asterisk-users] how to check the status of a channel   Alexander Lopez
• [asterisk-users] Linksys SPA-3000 Administration Guide   Marcos Rubino
• [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] Help with perl AGI script   Russell Bryant
• [asterisk-users] for some of my users, VoiceMail is being cutoff when leaving message   Randy Paries
• [asterisk-users] Asterisk and Siemens Legacy PBX   Jon Farmer
• [asterisk-users] g729 and trafic   Dovid Bender
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Avi Miller
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Armin Schindler
• [asterisk-users] Linksys SPA-3000 Administration Guide   Alberto Sagredo
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] Help with perl AGI script   Roy Kidder
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Armin Schindler
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] for some of my users, VoiceMail is being cutoff when leaving message   Doug Lytle
• [asterisk-users] Fax tone detected, but no fax extension for CAPI   Armin Schindler
• [asterisk-users] g729 and trafic   Michael Graves
• [asterisk-users] Using a DB for Configurations   Barry Fawthrop
• [asterisk-users] g729 and trafic   Michael Graves
• [asterisk-users] Using a DB for Configurations   Dovid Bender
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] need dialout help in python script   Shidan
• [asterisk-users] need dialout help in python script   shawn bright
• [asterisk-users] previous reload of asterisk did not finish   John covici
• [asterisk-users] Ring Groups   Chris Hembrow
• [asterisk-users] AG-168V not registering.   sdcharly at gmail.com
• [asterisk-users] Ring Groups   Hadley Rich
• [asterisk-users] Re: need a pointer regarding scripting asterisk   shawn bright
• [asterisk-users] Anyone use TACI.pl for a click to call app? - Doesn't seem to want to work for me   Christopher Aloi
• [asterisk-users] Variables sip redirects and call forward   C F
• [asterisk-users] Anyone use TACI.pl for a click to call app? - Doesn't seem to want to work for me   Mitchel Constantin
• [asterisk-users] [update] SIP/Qualify   Dovid Bender
• [asterisk-users] Variables sip redirects and call forward   Eric "ManxPower" Wieling
• [asterisk-users] How to "emulate" Music on Hold in a PHP AGI script?   Leo Burd
• [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] Asterisk and Siemens Legacy PBX   AstATN
• [asterisk-users] Help with perl AGI script   Russell Bryant
• [asterisk-users] How to "emulate" Music on Hold in a PHP AGI script?   Russell Bryant
• [asterisk-users] Variables sip redirects and call forward   C F
• [asterisk-users] HP ProLiant and Digium 24xxp   Robert Roach
• [asterisk-users] Variables sip redirects and call forward   Ira
• [asterisk-users] New people in this world and his problem with ISDN   Dominik Kiełb
• [SPAM] Re: [asterisk-users] SIP/Qualify   Pavel Jezek
• [asterisk-users] New people in this world and his problem with ISDN   Armin Schindler
• [asterisk-users] Video Conferencing over Asterisk   Siqhamo Sifo
• [asterisk-users] Ring Groups   Chris Hembrow
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] Video Conferencing over Asterisk   Olle E Johansson
• [asterisk-users] Asterisk and Siemens Legacy PBX   Wolfgang Zweimueller
• [SPAM] Re: [asterisk-users] SIP/Qualify   Olle E Johansson
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] Asterisk and Siemens Legacy PBX   Wolfgang Zweimueller
• [asterisk-users] Re: autocreatepeer in iax   Kamran Ahmad
• [asterisk-users] SER + Asterisk PSTN calls don't hung up   Ricardo Carvalho
• [asterisk-users] Re: autocreatepeer in iax   Kamran Ahmad
• [asterisk-users] Ztdummy - No audio in BackGround function   support_list
• [asterisk-users] How to "emulate" Music on Hold in a PHP AGI script?   Leo Burd
• [asterisk-users] Re: Load balancing of IAX2   Kamran Ahmad
• [SPAM] Re: [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] G729, IAX, polycom - trying to using 2 codecs   Dean at INKnBITs
• [asterisk-users] SIP/Qualify   Dovid Bender
• [asterisk-users] Re: Load balancing of IAX2   Florian Overkamp
• [asterisk-users] speech gaps with iax2   Pavel Jezek
• [asterisk-users] Inbound problems, no audio   Pablo Salvador Capo
• [asterisk-users] Caller ID problem on TDM400 FXO   Greg Delgado
• [asterisk-users] How to "emulate" Music on Hold in a PHP AGI script?   Tzafrir Cohen
• [asterisk-users] Re: [asterisk-dev] Questions regarding g.729 and g.711 in Asterisk   Rich Adamson
• [asterisk-users] Variables sip redirects and call forward   C F
• [asterisk-users] for some of my users, VoiceMail is being cutoff when leaving message   Randy Paries
• [asterisk-users] DTMF problems   Kohler, Jeffrey
• [asterisk-users] Ragi without rails possible ?   shawn bright
• [asterisk-users] Ragi without rails possible ?   Moises Silva
• [asterisk-users] DTMF problems   Moises Silva
• [asterisk-users] SER + Asterisk PSTN calls don't hung up   Ricardo Carvalho
• [asterisk-users] Conditional branching   bails
• [asterisk-users] Conditional branching   Doug Lytle
• [asterisk-users] Re: [asterisk-dev] Tuning Software Echo Cancellers   Rich Adamson
• [asterisk-users] Simple incomming fax solution ...   phil.dawson at marnock.com
• [asterisk-users] Stable asterisk version   phil.dawson at marnock.com
• [asterisk-users] Video Conferencing over Asterisk   Carlos Chavez
• [asterisk-users] Hotels...   Ken D'Ambrosio
• [asterisk-users] Simple incomming fax solution ...   Carlos Chavez
• [asterisk-users] Ragi without rails possible ?   shawn bright
• [asterisk-users] Fwd: * and GTalk testing   David Freeman
• [asterisk-users] Simple incomming fax solution ...   Marco Mouta
• [asterisk-users] Hotels...   Dovid Bender
• [asterisk-users] Ztdummy - No audio in BackGround function   Dovid Bender
• [asterisk-users] By week extension dialing   Dan Brummer
• [asterisk-users] looking to pay a consultant to help with my asterisk installation   Randy Paries
• [asterisk-users] Hotels...   Kristian Kielhofner
• [asterisk-users] Fwd: * and GTalk testing   Matt Riddell (NZ)
• [asterisk-users] Hotels...   Jonathan k. Creasy
• [asterisk-users] Hotels...   C F
• [asterisk-users] E1 for Voice and Data with MFC/R2   Carlos Chavez
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] RE: By week extension dialing   Brent Torrenga
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Eric "ManxPower" Wieling
• [asterisk-users] res_sqlite problems   Michael Iedema
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   C F
• [asterisk-users] Fwd: * and GTalk testing   David Freeman
• [asterisk-users] Re: [asterisk-dev] Questions regarding g.729 and g.711 in Asterisk   Olle E Johansson
• [asterisk-users] Video Conferencing over Asterisk   Olle E Johansson
• [asterisk-users] Asterisk and Siemens Legacy PBX   James Arscott
• [asterisk-users] Hotels...   Brian Vincent (C)
• [asterisk-users] By week extension dialing   Alexander Lopez
• [asterisk-users] RE: By week extension dialing   Dan Brummer
• [asterisk-users] Re: By week extension dialing   Steven
• [asterisk-users] By week extension dialing   Dan Brummer
• [asterisk-users] Meetme chat room with many users, and only 4 can talk, is there a max amount of users?   Manrique Feoli
• [asterisk-users] "Off-circuits are busy now. Please try your call again later"   Wolfgang Paul Rauchholz
• [asterisk-users] RE: By week extension dialing   John Williams
• [asterisk-users] Re: By week extension dialing   Dan Brummer
• [asterisk-users] DTMF problems   Kohler, Jeffrey
• [asterisk-users] By week extension dialing   Alexander Lopez
• [asterisk-users] MOH Silence   Douglas Garstang
• [asterisk-users] sip incoming stop working, what to look for in logs?   T. Shaw
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] DTMF problems   Moises Silva
• [asterisk-users] E1 for Voice and Data with MFC/R2   Moises Silva
• [asterisk-users] looking to pay a consultant to help with my asterisk installation   Moises Silva
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] RAM memory high Comsumption   Tzafrir Cohen
• [asterisk-users] How to link 2 existing calls   Leon Sun
• [asterisk-users] By week extension dialing   Dan Brummer
• [asterisk-users] FXS gateway/Channel Bank   Roger Workman
• [asterisk-users] FXS gateway/Channel Bank   Eric "ManxPower" Wieling
• [asterisk-users] MOH Silence   Wes Baehr
• [asterisk-users] Voicemail Platform   Roger Workman
• [asterisk-users] E1 for Voice and Data with MFC/R2   Carlos Chavez
• [asterisk-users] Re: Meetme chat room with many users, and only 4 can talk, is there a max amount of users?   Tony Mountifield
• [asterisk-users] voicemail in mp3 format   Ever Zalazar
• [asterisk-users] voicemail in mp3 format   Michiel van Baak
• [asterisk-users] Voicemail Platform   John Novack
• [asterisk-users] E1 for Voice and Data with MFC/R2   Moises Silva
• [asterisk-users] How to link 2 existing calls   Moises Silva
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] Voicemail Platform   Roger Workman
• [asterisk-users] Re: Meetme chat room with many users, and only 4 can talk, is there a max amount of users?   Manrique Feoli
• [asterisk-users] Re: Meetme chat room with many users, and only 4 can talk, is there a max amount of users?   Tony Mountifield
• [asterisk-users] SIP musicclass   Douglas Garstang
• [asterisk-users] NVFaxDetect and 1.2.10   John D. Coleman
• [asterisk-users] E1 for Voice and Data with MFC/R2   Jorge Cisneros
• [asterisk-users] Music On Hold Class Not Makin' Sense   Douglas Garstang
• [asterisk-users] E1 for Voice and Data with MFC/R2   Carlos Chavez
• [asterisk-users] sip incoming stop working, what to look for in logs?   T. Shaw
• [asterisk-users] sip incoming stop working, what to look for in logs?   Adrià Vidal
• [asterisk-users] sip incoming stop working, what to look for in logs?   T. Shaw
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] PROBLEM MUSIC ON HOLD   Elpidio Ramos
• [asterisk-users] PROBLEM MUSIC ON HOLD   Steve Kennedy
• [asterisk-users] (no subject)   Sony Veri Shandy
• [asterisk-users] agi script runs even if no answer   shawn bright
• [asterisk-users] agi script runs even if no answer   Alexander Lopez
• [asterisk-users] Polycom 301 and Linksys SRW224P PoE Switch   Santiago del Castillo
• [asterisk-users] Polycom 301 and Linksys SRW224P PoE Switch   BJ Weschke
• [asterisk-users] VoipNow 1.2.0 Beta   Greg Boehnlein
• [asterisk-users] agi script runs even if no answer   shawn bright
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   C F
• [asterisk-users] Polycom 1.6.7 Firmware Messages Button   Greg Boehnlein
• [asterisk-users] Hotels...   C F
• [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems   Greg Boehnlein
• Define PMS was: [asterisk-users] Hotels...   Alexander Lopez
• [asterisk-users] Hotels...   Bart Fisher
• [asterisk-users] problem- 0:10 long message   ismir saljic
• [asterisk-users] problem- 0:10 long message   Sharon Lim
• [asterisk-users] VoipNow 1.2.0 Beta   mitcheloc
• [asterisk-users] Re: agi script runs even if no answer   Tony Mountifield
• [asterisk-users] Conditional branching   bails
• [asterisk-users] Bluetooth phone as FXS/FXO with asterisk?   asterisk at anime.net
• [asterisk-users] Hotels...   Storm D. J. Petersen
• [Asterisk-Users] ISDN Y cable   Olivier
• [asterisk-users] Music On Hold Class Not Makin' Sense   Olle E Johansson
• [asterisk-users] Fwd: * and GTalk testing   Matt Riddell (NZ)
• [asterisk-users] set minimum iax jitterbuffer   Pavel Jezek
• [asterisk-users] VoipNow 1.2.0 Beta   Greg Boehnlein
• [asterisk-users] AGI doesn't execute PHP5 script   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] Problems with Codecs in Asterisk   Chan Kwang Mien
• [asterisk-users] set minimum iax jitterbuffer   Thomas Kenyon
• [asterisk-users] RE VoipNow 1.2.0 Beta   Matthew Warren
• [asterisk-users] set minimum iax jitterbuffer   Pavel Jezek
• [asterisk-users] RE VoipNow 1.2.0 Beta   Peter Bowyer
• [asterisk-users] codec_g729a.so coredump in SVN trunk   ast-lists at delvar.com
• [asterisk-users] RE VoipNow 1.2.0 Beta   Senad Jordanovic
• [asterisk-users] codec_g729a.so coredump in SVN trunk   Walt Reed
• [asterisk-users] AGI doesn't execute PHP5 script   Matt Riddell (NZ)
• [asterisk-users] set minimum iax jitterbuffer   Thomas Kenyon
• [asterisk-users] AGI doesn't execute PHP5 script   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] RE VoipNow 1.2.0 Beta   Greg Boehnlein
• [asterisk-users] IAX trunk behing NAT with dynamic IP   Andre Courchesne - Consultant
• SV: [asterisk-users] IAX trunk behing NAT with dynamic IP   Jon Schøpzinsky
• [asterisk-users] Re: agi script runs even if no answer   shawn bright
• [asterisk-users] Problems with Codecs in Asterisk   Rosli Sukri
• [asterisk-users] codec_g729a.so coredump in SVN trunk   Rich Adamson
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] AGI doesn't execute PHP5 script   Vahram Igityan
• [asterisk-users] IAX trunk behing NAT with dynamic IP   DM
• [asterisk-users] help with app_sms and chan_capi   Patrick Zwahlen
• [asterisk-users] Bluetooth phone as FXS/FXO with asterisk?   Tom Vile
• [asterisk-users] PROBLEM MUSIC ON HOLD   Zeeshan Zakaria
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   Doug Lytle
• [asterisk-users] Problems with Codecs in Asterisk   Dean at INKnBITs
• [asterisk-users] Festival Not Working   Zeeshan Zakaria
• [asterisk-users] Asterisk with BT's broadband voice service.   Ronan Mullally
• [asterisk-users] Jitterbuffer on SIP   Thierry Querette
• V: [asterisk-users] IAX trunk behing NAT with dynamic IP   Andre Courchesne - Consultant
• [asterisk-users] Re: agi script runs even if no answer   Tony Mountifield
• [asterisk-users] Problems with Codecs in Asterisk   Rosli Sukri
• [asterisk-users] Jitterbuffer on SIP   Patrick
• [asterisk-users] Problems with Codecs in Asterisk   Rich Adamson
• [asterisk-users] Problems with Codecs in Asterisk   Rosli Sukri
• [asterisk-users] Hotels...   Andrew Latham
• [asterisk-users] DTMF problems   Rosli Sukri
• [asterisk-users] Re: agi script runs even if no answer   shawn bright
• [asterisk-users] Problems with Codecs in Asterisk   Chan Kwang Mien
• [asterisk-users] Stopping Queue after nobody picked up the call . .   Ralph Liebessohn
• [asterisk-users] Probelm with IAX peers   David Freeman
• [asterisk-users] Sipura SPA-841   Giedrius Augys
• [asterisk-users] Cisco Phone Configuration Tool cannot find Files   Nate List
• [asterisk-users] Zaptel trunk failed to compile - Still but another error   Administrator TOOTAI
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   C F
• [asterisk-users] Asterisk and failover   Alex Lake
• [asterisk-users] Asterisk and failover   Alex Lake
• [asterisk-users] tellabs 2572 echo can and zaptel yellow alarm   marvin horst
• [asterisk-users] IAX trunk behing NAT with dynamic IP   Michiel van Baak
• [asterisk-users] A question about AGI and RECORD FILE   Randy Paries
• [asterisk-users] Polycom 1.6.7 firmware?   Louis-David Mitterrand
• [asterisk-users] Polycom 1.6.7 firmware?   Eric "ManxPower" Wieling
• [asterisk-users] Re: Polycom 1.6.7 firmware?   Louis-David Mitterrand
• [asterisk-users] Polycom 1.6.7 firmware?   Jessee J Holmes
• [asterisk-users] Re: A question about AGI and RECORD FILE   Tony Mountifield
• [asterisk-users] PRI Connection in Lima, Peru   Carlos Prieto
• [asterisk-users] Polycom 1.6.7 firmware?   Dean Collins
• [asterisk-users] Re: A question about AGI and RECORD FILE   Randy Paries
• [asterisk-users] Re: A question about AGI and RECORD FILE   Tony Mountifield
• [asterisk-users] polycom headset question   Dean Collins
• [asterisk-users] IAX Trunking Only ONE-WAY   Lic. Miguel Cavazos
• [asterisk-users] Re: Polycom 1.6.7 firmware?   Douglas Garstang
• [asterisk-users] polycom headset question   Bruce Reeves
• [asterisk-users] Host 207.174.202.2 failed to authenticate as teliax   Ryan Hayward
• [asterisk-users] polycom headset question   Dean Collins
• [asterisk-users] polycom headset question   Jessee J Holmes
• [asterisk-users] PRI Connection in Lima, Peru   Moises Silva
• [asterisk-users] polycom headset question   Evan P. Hall
• [asterisk-users] VoipNow 1.2.0 Beta   Daniel Salama
• [asterisk-users] PRI Connection in Lima, Peru   Carlos Prieto
• [asterisk-users] polycom headset question   Bill Gibbs
• [asterisk-users] PRI Connection in Lima, Peru   Eric "ManxPower" Wieling
• [asterisk-users] HFC-S Cards in the UK   Ron Wellsted
• [asterisk-users] polycom headset question   Evan P. Hall
• [asterisk-users] DTMF problems   Kohler, Jeffrey
• [asterisk-users] RE VoipNow 1.2.0 Beta   Miles Scruggs
• [asterisk-users] Possible to To Have Different Outgoing VM Messages, but One Mailbox?   Robert DeVries
• [asterisk-users] HFC-S Cards in the UK   Hermann Wecke
• [asterisk-users] HFC-S Cards in the UK   Ron Wellsted
• [asterisk-users] Possible to To Have Different Outgoing VM Messages, but One Mailbox?   Luki
• [asterisk-users] MP-124   Giedrius Augys
• [asterisk-users] Polycom 301 and Linksys SRW224P PoE Switch   Delca
• [asterisk-users] ${BLINDTRANSFER}->accountcode ?   Juan Pablo Abuyeres
• [asterisk-users] Bluetooth phone as FXS/FXO with asterisk?   asterisk at anime.net
• [asterisk-users] Bluetooth phone as FXS/FXO with asterisk?   Tom Vile
• [asterisk-users] Host 207.174.202.2 failed to authenticate as teliax   Jeremy McNamara
• [asterisk-users] Polycom 1.6.7 firmware?   Matt Florell
• [asterisk-users] ${BLINDTRANSFER}->accountcode ?   C F
• [asterisk-users] Sipura SPA-841   Dovid Bender
• [asterisk-users] ARA & Regseconds   Peder at NetworkOblivion
• [asterisk-users] Polycom 1.6.7 firmware?   Dean Collins
• [asterisk-users] Polycom 1.6.7 firmware?   Douglas Garstang
• [asterisk-users] Polycom 1.6.7 firmware?   Jessee J Holmes
• [asterisk-users] Polycom 1.6.7 firmware?   Dean Collins
• [asterisk-users] Polycom 1.6.7 firmware?   Avi Miller
• [asterisk-users] need dialout help in python script   Tzafrir Cohen
• [asterisk-users] HFC-S Cards in the UK   Chris Bagnall
• [asterisk-users] Polycom 1.6.7 firmware?   Stephen Murphy
• [asterisk-users] polycom config script   Dean Collins
• [asterisk-users] need dialout help in python script   shawn bright
• [asterisk-users] Polycom 1.6.7 firmware?   Avi Miller
• [asterisk-users] Polycom 1.6.7 firmware?   Eric "ManxPower" Wieling
• [asterisk-users] Polycom 1.6.7 firmware?   Stephen Murphy
• [asterisk-users] Sangoma A200D and DTMF Detection   Andres
• [asterisk-users] Handling inbound and outbound calls passed from a proxy   kjcsb
• [asterisk-users] Polycom 1.6.7 firmware?   Avi Miller
• [asterisk-users] hints causing hang in reload   Sean Cook
• [asterisk-users] Problems with Codecs in Asterisk   Rosli Sukri
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   Randall H.
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   Alex Robar
• [asterisk-users] AGI doesn't execute PHP5 script   Matt Riddell (NZ)
• [asterisk-users] AGI doesn't execute PHP5 script   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] realtime+mysql   Shaun
• [asterisk-users] realtime+mysql   Sharon Lim
• [asterisk-users] problem with queues   Muhammad Zeeshan Latif
• [asterisk-users] FW: problem with queues   Muhammad Zeeshan Latif
• [asterisk-users] G729   Khaled Chehab
• [asterisk-users] UK mobile "reject" codes   Julian Lyndon-Smith
• [asterisk-users] Re: realtime+mysql   Shaun
• [asterisk-users] Re: realtime+mysql   Sharon Lim
• [asterisk-users] cmd DIAL - Who picked up the call?   Kai Ober
• [asterisk-users] AGI doesn't execute PHP5 script [SOLVED]   Stefan-Michael. Guenther (in-put GbR)
• [asterisk-users] Two card NT-TE mode   support_list
• [asterisk-users] Asterisk and Siemens Legacy PBX   Wolfgang Zweimueller
• [asterisk-users] Re: [asterisk-dev] Questions regarding g.729 and g.711 in Asterisk   Thomas Kenyon
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   George Gardiner
• [asterisk-users] Two card NT-TE mode   Tzafrir Cohen
• [asterisk-users] Ever donate Software to Digium? If you did youra fool.   MBIT Technologies
• [asterisk-users] Re: Sangoma A200D and DTMF Detection   Rana Dutt
• [asterisk-users] AGI doesn't execute PHP5 script [SOLVED]   Matt Riddell (NZ)
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   Brian Capouch
• [asterisk-users] Ever donate Software to Digium? If you didyoura fool.   Bill Gibbs
• [asterisk-users] Phone Newbie Questions   Brian Becker
• [asterisk-users] Using a DB for Configurations   Barry Fawthrop
• [asterisk-users] Ever donate Software to Digium? If you did your afool.   Dean Collins
• [asterisk-users] Handling inbound and outbound calls passed from a proxy   Fran Oliveira
• [asterisk-users] Two card NT-TE mode   Klaus Darilion
• [asterisk-users] Ever donate Software to Digium? If you did your afool.   Alexander Lopez
• [asterisk-users] Phone Newbie Questions   Colin MacMillan
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Stephen G
• [asterisk-users] Jabber: Difference between client and component   Julian Lyndon-Smith
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Rich Adamson
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Stephen G
• [asterisk-users] Asterisk Configuration   R.Linga Reddy
• [asterisk-users] Asterisk Configuration   kritikus Araklidas
• [asterisk-users] Asterisk Configuration   Bruce Reeves
• [asterisk-users] Phone Newbie Questions   Tom Vile
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Dana Harding
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   Eric "ManxPower" Wieling
• [asterisk-users] Two card NT-TE mode   support_list
• [asterisk-users] Prague PTT?   Jason Aarons (US)
• [asterisk-users] ESCAUX releases net.PBX Free Edition   VOIP ESCAUX
• [asterisk-users] Deployment for less than 10 phones   Erick Perez
• [asterisk-users] e&m wink, TE110P, * answers too soon   Steve Linabery
• [asterisk-users] Tri-Link Technologies?   Don
• Autoreply: [asterisk-users] Deployment for less than 10 phones   gparlato at tnet.it
• Autoreply: [asterisk-users] e&m wink, TE110P, * answers too soon   gparlato at tnet.it
• [asterisk-users] How to, astcc and virtuemart   Julio Cáceres A.
• [asterisk-users] Snom MWI   J. Oquendo
• Autoreply: [asterisk-users] Tri-Link Technologies?   gparlato at tnet.it
• Autoreply: [asterisk-users] Tri-Link Technologies?   Don
• Autoreply: [asterisk-users] How to, astcc and virtuemart   gparlato at tnet.it
• Autoreply: [asterisk-users] Snom MWI   gparlato at tnet.it
• Autoreply: Re: Autoreply: [asterisk-users] Tri-Link Technologies?   gparlato at tnet.it
• [asterisk-users] Snom MWI   Colin Anderson
• Autoreply: [asterisk-users] Tri-Link Technologies?   David Freeman
• [asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2   Carlos Chavez
• Autoreply: [asterisk-users] Tri-Link Technologies?   Don
• [asterisk-users] RE: Ever donate Software to Digium? If you did your afool.   Yaakov Menken
• [asterisk-users] High Availability   Eric Rousse
• [asterisk-users] BriStuff | HFC-S | Progress() | Early B3 on incoming calls from PSTN   Stefan Gofferje
• [asterisk-users] HFC-S Cards in the UK   Ron Wellsted
• [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?   Manrique Feoli
• [asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2   Moises Silva
• [asterisk-users] Re: Sipura SPA-3000 vs Sangoma A200   David Cook
• [asterisk-users] DTMF codes in feature.conf not comming through   Henrik Ostergaard Madsen
• [asterisk-users] Ignoring the # key on a call   patrick
• [asterisk-users] wildcard always busy   Ralph Liebessohn
• [asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2   Thierry Querette
• [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?   Dean Collins
• Autoreply: [asterisk-users] Snom MWI   Austin Denyer
• Autoreply: [asterisk-users] Snom MWI   Andrew D Kirch
• [asterisk-users] BriStuff | HFC-S | Progress() | Early B3 on incoming calls from PSTN   Matteo Brancaleoni
• Autoreply: [asterisk-users] Snom MWI   Andrew D Kirch
• [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?   Rich Adamson
• [asterisk-users] Force peer source ip   Robin Ericsson
• [asterisk-users] Ignoring the # key on a call   Matt Riddell (NZ)
• [asterisk-users] Ignoring the # key on a call   Alex Robar
• Autoreply: [asterisk-users] Snom MWI   Rich Adamson
• [asterisk-users] Re: Ever donate Software to Digium? If you did your afool.   Gonzalo Servat
• [asterisk-users] Re: Ever donate Software to Digium? If you didyour afool.   Don
• [asterisk-users] Mac Address Authentication Methods   Daniel Botelho P. Moraes
• [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?   Manrique Feoli
• [asterisk-users] Jitterbuffer on SIP   Thierry Querette
• [asterisk-users] HFC-S Cards in the UK   Wolfgang Zweimueller
• [asterisk-users] Re: Snom MWI   Benny Amorsen
• [asterisk-users] Jitterbuffer on SIP   Jan Fousek
• [asterisk-users] Mac Address Authentication Methods   Rich Adamson
• [asterisk-users] can Digium FXS channels support been half mile to 1 mile length away from phone?   C F
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   C F
• [asterisk-users] Re: Sangoma A200D and DTMF Detection   Andres
• [asterisk-users] How many digits are collected   Bruce Ferrell
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   Michael Graves
• [asterisk-users] How many digits are collected   C F
• [asterisk-users] Ever donate Software to Digium? If you did your a fool.   Matt Florell
• [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole   Tom
• [asterisk-users] Callback and Asterisks   Vic
• [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole   Mr. Jones
• [asterisk-users] Callback and Asterisks   Hermann Wecke
• [asterisk-users] SIP trunks: order or type   Shaun Hofer
• [asterisk-users] Re: Re: realtime+mysql   Shaun
• [asterisk-users] Iptables ,rtp   Siqhamo Sifo
• [asterisk-users] transfer call von D-channel   Christophorus Laube
• [asterisk-users] Iptables ,rtp   Michiel van Baak
• [asterisk-users] MixMonitor stops after attended call transfer   Haspers
• [asterisk-users] transfer call von D-channel   Koopmann, Jan-Peter
• [asterisk-users] Asterisk Load Balance   Andy Chung (Power-All)
• [asterisk-users] Sangoma A101 problem   Marcin Kwiatkowski
• [asterisk-users] SIP trunks: order or type   Marco Mouta
• [asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2   Steve Underwood
• [asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2   Steve Underwood
• [asterisk-users] Clipped audio at beginning of SIP calls.   Martin Joseph
• [asterisk-users] Re: can Digium FXS channels support been half mileto 1 mile length away from phone?   Steven
• [asterisk-users] Realtime SIP Authentication   ronn100200 at aim.com
• [asterisk-users] MFC/R2 country and carrier specific protocol variants   Steve Underwood
• [asterisk-users] Realtime SIP Authentication   Rushowr
• [asterisk-users] Help with newbie: D-link admin setup   Dominic Son
• [asterisk-users] Asterisk Configuration   R.Linga Reddy
• [asterisk-users] Question on iax2 show netstats   Matt
• [asterisk-users] e&m wink, TE110P, * answers too soon   Sean Cook
• [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole   Shaw Terwilliger
• [asterisk-users] Asterisk Configuration   Jordi Nelissen
• [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk   M D
• [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk   Kevin Savoy
• [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk= same in 1.2.10   WideVOIP
• [asterisk-users] can i detect a voice with asterisk ?   shawn bright
• [asterisk-users] Sipura SPA-3000 vs Sangoma A200   Stephen G
• [asterisk-users] Integrating Asterisk with a Panasonic D500 using MFC/R2   Thierry Querette
• [asterisk-users] Samsung Prostar DCS   Curt Shaffer
• [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk   M D
• [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole   Don Tasker
• [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk   John covici
• [asterisk-users] Set DID?   Dean Collins
• [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole   dorn hetzel
• [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole   C F
• [asterisk-users] can i detect a voice with asterisk ?   Ira
• [asterisk-users] Ever donate Software to Digium? If you did youra fool.   shadowym
• [asterisk-users] Ever donate Software to Digium? If you did youra fool.   Steve Underwood
• [asterisk-users] Samsung Prostar DCS   Florian Overkamp
• [asterisk-users] sms callback?   B
• [asterisk-users] can i detect a voice with asterisk ?   Eric "ManxPower" Wieling
• [asterisk-users] Set DID?   Eric "ManxPower" Wieling
• [asterisk-users] Odd IAX stats   Curt Shaffer
• [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk   Kevin Savoy
• [asterisk-users] Asterisk Voicemail Setup   Wasif
• [asterisk-users] sms callback?   Rene Kluwen
• [asterisk-users] Re: sms callback?   B
• [asterisk-users] Cisco Buddies   Peder at NetworkOblivion
• [asterisk-users] Asterisk Voicemail Setup   Alex Robar
• [asterisk-users] Set DID?   Dean Collins
• [asterisk-users] IAXy can't connect to analog phone   Martti Tienhaara
• [asterisk-users] Snakes On A Plane using Asterisk?   Hugh L. Johnson
• [asterisk-users] Cisco Buddies   Adrià Vidal
• [asterisk-users] Record phone call using asterisk   Sam Tam
• [asterisk-users] can i detect a voice with asterisk ?   shawn bright
• [asterisk-users] Set DID?   Eric "ManxPower" Wieling
• [asterisk-users] Snakes On A Plane using Asterisk?   Tom Vile
• [asterisk-users] Set DID?   Marco Mouta
• [asterisk-users] Set DID?   Dean Collins
• [asterisk-users] Set DID?   Dean Collins
• [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group - Saturday August 12th - 11:30am   asterisk_help at iwishi.nu
• [asterisk-users] can i detect a voice with asterisk ?   Jerry Geis
• [asterisk-users] can i detect a voice with asterisk ?   shawn bright
• [asterisk-users] can i detect a voice with asterisk ?   Matt
• [asterisk-users] Cisco Buddies   Peder at NetworkOblivion
• [asterisk-users] Cisco Buddies   Peder at NetworkOblivion
• [asterisk-users] A good price for FXS 48 ports?   Javier Matos Odut
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   hugolivude
• [asterisk-users] A good price for FXS 48 ports?   Thierry Querette
• [asterisk-users] A good price for FXS 48 ports?   Don Tasker
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Avi Miller
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   C F
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Don
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   C F
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Alex Robar
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Alex Robar
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   C F
• [asterisk-users] Cisco Buddies   Adrià Vidal
• [asterisk-users] Re: sms callback?   Rene Kluwen
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   hugolivude
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Don
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Don
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   Don
• [asterisk-users] SIP trunks: order or type   Shaun Hofer
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   hugolivude
• [asterisk-users] Correct syntax for Set(CALLERID(all)...   C F
• [asterisk-users] SIP trunks: order or type   Rich Adamson
• [asterisk-users] Quick One - PHP Script to restart Asterisk   Corporate IT Solutions - Michael Dunne
• [asterisk-users] Phone number lookup public database   Thameem Ansari
• [asterisk-users] No path to translate from SIP/30-09df8928(4) to SIP/3470075XXXX-09dfb518(256)   Wolfgang Paul Rauchholz
• [asterisk-users] No path to translate from SIP/30-09df8928(4) to SIP/3470075XXXX-09dfb518(256)   Jeremy McNamara
• [asterisk-users] Re: Load balancing of IAX2   Kamran Ahmad
• [asterisk-users] Re: Load balancing of IAX2   Kamran Ahmad
• [asterisk-users] No path to translate from SIP/30-09df8928(4) to SIP/3470075XXXX-09dfb518(256)   Avi Miller
• [asterisk-users] No path to translate from SIP/30-09df8928(4) to SIP/3470075XXXX-09dfb518(256)   Hadley Rich
• [asterisk-users] Quick One - PHP Script to restart Asterisk   Paul Hales
• [asterisk-users] SIP trunks: order or type   Fran Oliveira
• [asterisk-users] question about oh323 and ring tone   asterisk at frameweb.it
• [asterisk-users] where/when to set__TRANSFER_CONTEXT ?   Kai Ober
• [asterisk-users] Port Forwarding SIP rtp   Siqhamo Sifo
• [asterisk-users] High Availability with PRI failover   Vicente Aguilar
• [asterisk-users] High Availability with PRI failover   Senad Jordanovic
• [asterisk-users] In CDR record not what I want   Matthias Fechner
• [asterisk-users] In CDR record not what I want   Rushowr
• [asterisk-users] Port Forwarding SIP rtp   Rosli Sukri
• [asterisk-users] Port Forwarding SIP rtp   Peter Bowyer
• [asterisk-users] Asterisk GUI tool needed   Rizwan Hisham
• [asterisk-users] Polycom just disconnects   Bartosz Jozwiak
• [asterisk-users] Asterisk IAXmodem HylaFax?   Damon Estep
• [asterisk-users] Port Forwarding SIP rtp   Evalyn Wafula
• [asterisk-users] Asterisk GUI tool needed   Tijl Van den Broeck
• [asterisk-users] Polycom just disconnects   Damon Estep
• [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution &nbs