April 2006 Archives by date

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Starting: Sat Apr 1 00:01:45 MST 2006
Ending: Sun Apr 30 23:58:37 MST 2006
Messages: 3610

• [Asterisk-Users] ooh323 and g729 - any issue?   voipman
• [Asterisk-Users] voicemail to email sending problems   Jordan Novak
• [Asterisk-Users] Quintum Tenor DX4060   Steve Totaro
• [Asterisk-Users] Asterisk Referral - Cleanup on Aisle 7   Steve Totaro
• [Asterisk-Users] How to use Sendtxt?   Ronald Wiplinger
• [Asterisk-Users] OT: Polycom IP501 and Speed Dials   Avi Miller
• [Asterisk-Users] kernel recompilation on a asterisk server   nik600
• [Asterisk-Users] Dial cmd has delay for the last dialed number on FXO   Franz Wu
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Lee Archer
• [Asterisk-Users] kernel recompilation on a asterisk server   Marco Campos
• [Asterisk-Users] kernel recompilation on a asterisk server   nik600
• [Asterisk-Users] How to check if a phone / line is used?   Ronald Wiplinger
• [Asterisk-Users] INX (Internationalnumber.com)   voipman
• [Asterisk-Users] Newbie question - sip.conf incoming contexts   Steve Gladden
• [Asterisk-Users] Panasonic KXTD 1232 6   Krzysztof Drewicz
• [Asterisk-Users] Echo cancellation problem   Sergio Chersovani
• [Asterisk-Users] Asterisk box with unreliable ping/latency   Bjorn O
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Melcon Moraes
• [Asterisk-Users] Re: How is Teliax ?   Rich Adamson
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Matt
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Lee Archer
• [Asterisk-Users] Confused on Agents and Queues   Matt
• [Asterisk-Users] Refer Accountcode Bug   Elton Machado
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Lee Archer
• [Asterisk-Users] Asterisk box with unreliable ping/latency   Rich Adamson
• [Asterisk-Users] Newbie question - sip.conf incoming contexts   Rich Adamson
• [Asterisk-Users] Dial cmd has delay for the last dialed number on FXO   Rich Adamson
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Matt
• [Asterisk-Users] Routing SIP calls via URI   Shad Mortazavi
• [Asterisk-Users] Free Software/Open Source Telephony-Summit 2006   Kevin P. Fleming
• [Asterisk-Users] AGI hangup problem   Branko Samardzic
• [Asterisk-Users] Problem: ringtones stop unexpectedly when multiple channels are dialed   Carlos A. Alfaro
• [Asterisk-Users] TO have ringing tone instead MOH   Alberto Sagredo
• [Asterisk-Users] Problem: ringtones stop unexpectedly when multiple channels are dialed   Julian J. M.
• [Asterisk-Users] AGI hangup problem   Stefan Reuter
• [Asterisk-Users] Incorrect CDR results   Michael Welter
• [Asterisk-Users] *.conf generator   mustardman29
• [Asterisk-Users] Incorrect CDR results   Andrew Kohlsmith
• [Asterisk-Users] Iaxmodem speed limit?   Lee Howard
• [Asterisk-Users] AGI hangup problem   Branko Samardzic
• [Asterisk-Users] TO have ringing tone instead MOH   Matt
• [Asterisk-Users] *.conf generator   Matt
• [Asterisk-Users] *.conf generator   Matt
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Lee Archer
• [Asterisk-Users] TO have ringing tone instead MOH   Melcon Moraes
• [Asterisk-Users] no audio   Luis herrera
• [Asterisk-Users] chan-capi: Sending digits on a bri (isdn) d-channel   Raoul Bönisch
• [Asterisk-Users] Re: no audio   hugolivude
• [Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)   Rich Adamson
• [Asterisk-Users] OT: Polycom IP501 and Speed Dials   Jeff Herring
• [Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)   Kevin P. Fleming
• [Asterisk-Users] G729 codec problems   RumaTech
• [Asterisk-Users] channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8446b50', 10 retries!   Il Neofita
• [Asterisk-Users] vmail access problem   Ever Zalazar
• [Asterisk-Users] G729 codec problems   Kevin P. Fleming
• [Asterisk-Users] G729 codec problems   Rudolf Ladyzhenskii
• [Asterisk-Users] Re: no audio   Tom Vile
• [Asterisk-Users] FreePBX on Debian   Christian Gröger
• [Asterisk-Users] H323 on way voice   Il Neofita
• [Asterisk-Users] H323 on way voice   isamar at isamarmaia.org
• [Asterisk-Users] Problem: ringtones stop unexpectedly   Carlos A. Alfaro
• [Asterisk-Users] Zap channels - help   Tzafrir Cohen
• [Asterisk-Users] H323 on way voice   Jeremy McNamara
• [Asterisk-Users] Panasonic KXTD 1232 6   C F
• [Asterisk-Users] voicemail to email sending problems   C F
• [Asterisk-Users] *.conf generator   mustardman29
• [Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)   Rich Adamson
• [Asterisk-Users] Problem: ringtones stop unexpectedly   Alyed Tzompa
• [Asterisk-Users] G729 codec problems   Alyed Tzompa
• [Asterisk-Users] Re: no audio   Alyed Tzompa
• [Asterisk-Users] G729 Passthrough question   From PH
• [Asterisk-Users] G729 codec problems   Kevin P. Fleming
• [Asterisk-Users] Re: How is Teliax ?   asterisk at anime.net
• [Asterisk-Users] H323 on way voice   gsalas
• [Asterisk-Users] Newbie question - sip.conf incoming contexts   Steve Gladden
• [Asterisk-Users] morcdr v0.1 released   Mindaugas Kezys
• [Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)   Tim Panton
• [Asterisk-Users] Cisco 7960 nat problems.   Shaun
• [Asterisk-Users] H323 on way voice   Il Neofita
• [Asterisk-Users] FreePBX on Debian   stoffell
• [Asterisk-Users] H323 on way voice   isamar at isamarmaia.org
• [Asterisk-Users] Line Pick up Problem   Mr Asterisk
• [Asterisk-Users] polycom overlap dialing?   asterisk at anime.net
• [Asterisk-Users] polycom overlap dialing?   Noah Miller
• [Asterisk-Users] OT: Polycom IP501 and Speed Dials   Noah Miller
• [Asterisk-Users] Building Asterisk embedded device   Noah Miller
• [Asterisk-Users] Asterisk box with unreliable ping/latency   Noah Miller
• [Asterisk-Users] Cisco 7960 nat problems.   Doug Lytle
• [Asterisk-Users] G729 codec problems   Rich Adamson
• [Asterisk-Users] Re: How is Teliax ?   Rich Adamson
• [Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)   Rich Adamson
• [Asterisk-Users] Cisco 7960 nat problems.   Rich Adamson
• [Asterisk-Users] H323 on way voice   Balgansuren Batsukh
• [Asterisk-Users] vmail access problem   Kyle Sexton
• [Asterisk-Users] Newbie question - sip.conf incoming contexts   Rich Adamson
• [Asterisk-Users] DID registration status   Giridhar Reddy Bandi
• [Asterisk-Users] DID registration status   Rich Adamson
• [Asterisk-Users] DID registration status   Kyle Sexton
• [Asterisk-Users] H323 on way voice   Kyle Sexton
• [Asterisk-Users] Asterisk 2.0 Where to download   Nguyen Trung Tin
• [Asterisk-Users] Asterisk 2.0 Where to download   Peter Bowyer
• [Asterisk-Users] Asterisk 2.0 Where to download   Kristian Kielhofner
• [Asterisk-Users] Re: caller anounce   Miles Scruggs
• [Asterisk-Users] G729 codec problems   Steve Kennedy
• [Asterisk-Users] Asterisk 2.0 Where to download   Doug Lytle
• [Asterisk-Users] FreePBX on Debian   Christian Gröger
• [Asterisk-Users] no audio between sip channels * 1.2.6   John Millican
• [Asterisk-Users] no audio   Dovid Bender
• [Asterisk-Users] G729 codec problems   Kevin P. Fleming
• [Asterisk-Users] Re: caller anounce   Dovid Bender
• [Asterisk-Users] Re: caller anounce   Miles Scruggs
• [Asterisk-Users] Asterisk 2.0 Where to download   Dovid Bender
• [Asterisk-Users] Zap channels - help   Josué Conti
• [Asterisk-Users] Asterisk hosted solution   Dovid Bender
• [Asterisk-Users] Blacklist out bound numbers from file   Dovid Bender
• [Asterisk-Users] Asterisk 2.0 Where to download   Kerry Garrison
• [Asterisk-Users] Asterisk 2.0 Where to download   Dovid Bender
• [Asterisk-Users] incoming triggers seperate outbound   Dovid Bender
• [Asterisk-Users] Re: Cisco 7960 nat problems.   Shaun Reitan
• [Asterisk-Users] Re: Cisco 7960 nat problems.   Shaun Reitan
• [Asterisk-Users] can automon work with MixMonitor   Franz Wu
• [Asterisk-Users] Asterisk "answering machine" replacement, "WaitForRing()", application return values   smc+astuser at dogphilosophy.net
• [Asterisk-Users] Subversion mirrors of Asterisk, Zaptel and libpri rebuilt   Kevin P. Fleming
• [Asterisk-Users] Re: Cisco 7960 nat problems.   Doug Lytle
• [Asterisk-Users] Connecting Asterisk to "traditional" phone central   Andrew Nowrot
• [Asterisk-Users] Newbie question - sip.conf incoming contexts   Marco Mouta
• [Asterisk-Users] Connecting Asterisk to "traditional" phone central   Steve Totaro
• [Asterisk-Users] Voicemail() - Reading exit or return results   Bart Fisher
• [Asterisk-Users] G729 codec problems   Rudolf Ladyzhenskii
• [Asterisk-Users] Codec Problem   Il Neofita
• [Asterisk-Users] Information about LOCAL/ Channel   Stuart Elvish - Dallas Delta Corporation Pty Ltd
• [Asterisk-Users] Information about LOCAL/ Channel   C F
• [Asterisk-Users] Re: caller anounce   Tom Vile
• [Asterisk-Users] Re: caller anounce   Tom Vile
• [Asterisk-Users] Re: caller anounce   Miles Scruggs
• [Asterisk-Users] Newbie question - sip.conf incoming contexts   Tom Vile
• [Asterisk-Users] Re: caller anounce   Tom Vile
• [Asterisk-Users] Reporting?   Doug Geary
• [Asterisk-Users] Who is on a call?   Ronald Wiplinger
• [Asterisk-Users] Re: caller anounce   Miles Scruggs
• [Asterisk-Users] Re: caller anounce   Steven Job
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Heidi Mendoza
• [Asterisk-Users] ZapBarge but ability to talk to the agent   Andre Courchesne - Consultant
• [Asterisk-Users] H323 on way voice   Jeremy McNamara
• [Asterisk-Users] Who is on a call?   Douglas Garstang
• [Asterisk-Users] H323 on way voice   Jeremy McNamara
• [Asterisk-Users] Re: caller anounce   Miles Scruggs
• [Asterisk-Users] ASTCC: How to reset "in-use" flag automatically ?   Ronald Wiplinger
• [Asterisk-Users] Connecting Asterisk to "traditional" phone central   Andrew Nowrot
• [Asterisk-Users] cannot set outgoing cid   Sebastian Reitenbach
• [Asterisk-Users] Codec Problem   stevanus
• [Asterisk-Users] Re: internals and ISDN calls fail when Internet is down   Tomislav Parčina
• [Asterisk-Users] Re: Building Asterisk embedded device   Tomislav Parčina
• [Asterisk-Users] Re: oh323 - unable to install   Tomislav Parčina
• [Asterisk-Users] Re: ooh323 and g729 - any issue?   Tomislav Parčina
• [Asterisk-Users] Re: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8446b50', 10 retries!   Tomislav Parčina
• [Asterisk-Users] Re: Building Asterisk embedded device   Lee Chit Seong
• [Asterisk-Users] Comparison of Business Edition VS Open Source   Lilantha Karunaratne
• [Asterisk-Users] Re: Junghanns and Digium TDM400?   Koopmann, Jan-Peter
• [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.   Koopmann, Jan-Peter
• [Asterisk-Users] Re: Compatible Asterisk Connectivity Cards : Sangoma   Tomislav Parčina
• [Asterisk-Users] Re: Asterisk hosted solution   Thorben Jensen
• [Asterisk-Users] Re: Re: Cisco 7960 nat problems.   Shaun
• [Asterisk-Users] Problem: ringtones stop unexpectedly   Carlos A. Alfaro
• [Asterisk-Users] Re: Compatible Asterisk Connectivity Cards : Sangoma   Matt Florell
• [Asterisk-Users] Who is on a call?   Peter Bowyer
• [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Isaac Xiao
• [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Matt Florell
• [Asterisk-Users] chan-capi: Sending digits on a bri (isdn) d-channel   Armin Schindler
• [Asterisk-Users] No CDR in Macro after Dial   Arjan Kroon
• [Asterisk-Users] Diva Server BRI echo options   Giuseppe
• [Asterisk-Users] cannot set outgoing cid   Sebastian Reitenbach
• [Asterisk-Users] Diva Server BRI echo options   Avi Miller
• [Asterisk-Users] Diva Server BRI echo options   Giuseppe
• [Asterisk-Users] R2 protocol error   Dennis Nacino
• [Asterisk-Users] G729 codec problems   Steve Kennedy
• [Asterisk-Users] Re: How to use Sendtxt?   Dennis Nacino
• [Asterisk-Users] Diva Server BRI echo options   Avi Miller
• [Asterisk-Users] bristuff for * 1.2.6/zaptel 1.2.5   DRi at b-w-computer.de
• [Asterisk-Users] update asterisk in a production system   nik600
• [Asterisk-Users] bristuff for * 1.2.6/zaptel 1.2.5   Benoit Panizzon
• [Asterisk-Users] Diva Server BRI echo options   Armin Schindler
• [Asterisk-Users] bristuff for * 1.2.6/zaptel 1.2.5   DRi at b-w-computer.de
• [Asterisk-Users] Annonuce Me Feature   Dovid Bender
• [Asterisk-Users] Bad Pick up line   Mr Asterisk
• [Asterisk-Users] Asterisk "answering machine" replacement, "WaitForRing()", application return values   Dovid Bender
• [Asterisk-Users] Diva Server BRI echo options (fixed)   Giuseppe
• [Asterisk-Users] call transfer to external phone number   Giuseppe
• [Asterisk-Users] bristuff for * 1.2.6/zaptel 1.2.5   stoffell
• [Asterisk-Users] call transfer to external phone number   Dovid Bender
• [Asterisk-Users] Re: caller anounce   Dovid Bender
• [Asterisk-Users] R2 protocol error   Melcon Moraes
• [Asterisk-Users] ZapBarge but ability to talk to the agent   Dovid Bender
• [Asterisk-Users] AMILogin and case sensitive   richard Coco
• [Asterisk-Users] Building Asterisk embedded device   Quark IT - Hilton Travis
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Quark IT - Hilton Travis
• [Asterisk-Users] Who is on a call?   Peter Fern
• [Asterisk-Users] bristuff for * 1.2.6/zaptel 1.2.5   Benoit Panizzon
• [Asterisk-Users] R2 protocol error   Steve Underwood
• [Asterisk-Users] Re: Compatible Asterisk Connectivity Cards : Sangoma   Rich Adamson
• [Asterisk-Users] Concurrent calls to voipstunt and other providers   Ronald Wiplinger
• [Asterisk-Users] update asterisk in a production system   Aaron Daniel
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Brian Roy
• [Asterisk-Users] Re: Re: Compatible Asterisk Connectivity Cards : Sangoma   Tomislav Parčina
• [Asterisk-Users] G729 codec problems   Rudolf Ladyzhenskii
• [Asterisk-Users] Frustrated with echo...   Steve Jones
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Pete Barnwell
• [Asterisk-Users] call transfer to external phone number   C F
• [Asterisk-Users] Callback auto dialing   Cosmin Prund
• [Asterisk-Users] call transfer to external phone number   Cosmin Prund
• [Asterisk-Users] Coice recognition IVR?   Cosmin Prund
• [Asterisk-Users] Re: ooh323 and g729 - any issue?   Erick Perez
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Matt
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Steve Underwood
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Lee Archer
• [Asterisk-Users] Coice recognition IVR?   Cristian Draghici
• [Asterisk-Users] Re: Re: How is Teliax ?   Steven
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Matt Florell
• [Asterisk-Users] Re: Who is on a call?   Steven
• [Asterisk-Users] Random music not so 'random'   Matt
• [Asterisk-Users] Re: Random music not so 'random'   Matt
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Andre Ruiz
• [Asterisk-Users] SIP Responsecodes   Freddi Hansen
• [Asterisk-Users] Hinting   Aaron Daniel
• [Asterisk-Users] Coice recognition IVR?   Cosmin Prund
• [Asterisk-Users] call transfer to external phone number   C F
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Lee Archer
• [Asterisk-Users] Beginner: PBX for my house   erik
• [Asterisk-Users] Anybody success using Asterisk 1.2.6 and SpanDSP 0.0.3 pre 6?   Pimjai Wesnarat
• [Asterisk-Users] web meetme instructions   Richard OSS
• [Asterisk-Users] Beginner: PBX for my house   Cory Andrews
• [Asterisk-Users] Beginner: PBX for my house   Kerry Garrison
• [Asterisk-Users] call transfer to external phone number   Keith Geffert
• [Asterisk-Users] Confused on Agents and Queues   Alan Ferrency
• [Asterisk-Users] Pickup() h323   Pavel Jezek
• [Asterisk-Users] Callback auto dialing   Tim Panton
• [Asterisk-Users] Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)   Erick Perez
• [Asterisk-Users] Building Asterisk embedded device   mustardman29
• [Asterisk-Users] Re: Re: Compatible Asterisk Connectivity Cards :Sangoma   mustardman29
• [Asterisk-Users] Beginner: PBX for my house   RandyW
• [Asterisk-Users] Anybody success using Asterisk 1.2.6 and Spa nDSP 0.0.3 pre 6?   Colin Anderson
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Andre Ruiz
• [Asterisk-Users] Unable to connect to remote asterisk (does / var/run/asterisk.ctl exist?)   Colin Anderson
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Andrew Kirch
• [Asterisk-Users] Pickup() h323   Joshua Colp
• [Asterisk-Users] Beginner: PBX for my house   Kerry Garrison
• [Asterisk-Users] Who is on a call?   Joshua Colp
• [Asterisk-Users] Hinting   Kevin P. Fleming
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Joshua Colp
• [Asterisk-Users] SIP Responsecodes   Joshua Colp
• [Asterisk-Users] Coice recognition IVR?   Joshua Colp
• [Asterisk-Users] Anybody success using Asterisk 1.2.6 and Spa nDSP 0.0.3 pre 6?   Pimjai Wesnarat
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   mustardman29
• [Asterisk-Users] Re: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)   Steven
• [Asterisk-Users] Is it a stun problem: 63 to 1800 msec   Ronald Wiplinger
• [Asterisk-Users] Re: Re: Compatible Asterisk Connectivity Cards :Sangoma   Heidi Mendoza
• [Asterisk-Users] SIP Responsecodes   Douglas Garstang
• [Asterisk-Users] Coice recognition IVR?   Cosmin Prund
• [Asterisk-Users] Unable to connect to remote asterisk (does / var/run/asterisk.ctl exist?)   Erick Perez
• [Asterisk-Users] SIP Responsecodes   Roger Schreiter
• [Asterisk-Users] SIP Responsecodes   Joshua Colp
• [Asterisk-Users] Hinting   Aaron Daniel
• [Asterisk-Users] Maximum duration of Voicemail messages   Andy Tan
• [Asterisk-Users] Pickup() h323   Pavel Jezek
• [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Matt Roth
• [Asterisk-Users] Unable to connect to remote asterisk (does / var/run/asterisk.ctl exist?)   Erick Perez
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Koopmann, Jan-Peter
• [Asterisk-Users] Coice recognition IVR?   Vamsi Pottangi
• [Asterisk-Users] Stupid newbie question   Galley, Mark
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Matt Roth
• [Asterisk-Users] Stupid newbie question   Tzafrir Cohen
• [Asterisk-Users] Re: How is Teliax ?   asterisk at anime.net
• [Asterisk-Users] Frustrated with echo...   asterisk at anime.net
• [Asterisk-Users] Critical Transaction failed: Client non-INVITE - SJPHONE connected to Asterisk   Marco Mouta
• [Asterisk-Users] Interrupting a call   Eric Jacksch
• [Asterisk-Users] Re: How is Teliax ?   Kristian Kielhofner
• [Asterisk-Users] Lockups after Asterisk upgrade   Patrick Staehlin
• [Asterisk-Users] Hinting   Kevin P. Fleming
• [Asterisk-Users] Stupid newbie question   Galley, Mark
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Kevin P. Fleming
• [Asterisk-Users] Hinting   Aaron Daniel
• [Asterisk-Users] web meetme instructions   Dan Austin
• [Asterisk-Users] Hardware question about Redfone's foneBridge   Asterisk Consulting
• [Asterisk-Users] Stupid newbie question   Colin Anderson
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Matt
• [Asterisk-Users] Hinting   Anthony Rodgers
• [Asterisk-Users] Anybody know about Cisco VOIP routers?   Doug
• [Asterisk-Users] Hinting   Sean Cook
• [Asterisk-Users] 411 Directory: First, Last or Both?   Doug
• [Asterisk-Users] Stupid newbie question   Galley, Mark
• [Asterisk-Users] Coice recognition IVR?   wendell hamilton
• [Asterisk-Users] Hinting   Aaron Daniel
• [Asterisk-Users] Queues - Dumb question   Wes Baehr
• [Asterisk-Users] Hardware question about Redfone's foneBridge   Bruce Reeves
• [Asterisk-Users] polycom overlap dialing?   Jerry Jones
• [Asterisk-Users] Hinting   Sean Cook
• [Asterisk-Users] Inter-Asterisk SIP and CalleriID   Adam Robins
• [Asterisk-Users] Frustrated with echo...   Mike Dent
• [Asterisk-Users] Beginner: PBX for my house   Dovid Bender
• [Asterisk-Users] Beginner: PBX for my house   erik
• [Asterisk-Users] Confused on Agents and Queues   Matt
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Matt Roth
• [Asterisk-Users] Pickup() h323   Jeremy McNamara
• [Asterisk-Users] Hinting   Darrick Hartman
• [Asterisk-Users] Asterisk compiling problems...   George A. Michalopoulos
• [Asterisk-Users] Anybody know about Cisco VOIP routers?   Greg Oliver
• [Asterisk-Users] polycom overlap dialing?   C F
• [Asterisk-Users] web meetme   Jordan Novak
• [Asterisk-Users] polycom overlap dialing?   Anton Krall
• [Asterisk-Users] applicationmap   Anton Krall
• [Asterisk-Users] Meetme admin   Jordan Novak
• [Asterisk-Users] Anybody know about Cisco VOIP routers?   Bill Gibbs
• [Asterisk-Users] Meetme admin   Anton Krall
• [Asterisk-Users] warnings during parsing of misdn.conf   Sébastien Mortier
• [Asterisk-Users] web meetme   Dan Austin
• [Asterisk-Users] applicationmap   C F
• [Asterisk-Users] applicationmap   Anton Krall
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Matt
• [Asterisk-Users] Need to Install Fax to Email feature   Wasif
• [Asterisk-Users] Hinting   Anthony Rodgers
• [Asterisk-Users] polycom overlap dialing?   asterisk at anime.net
• [Asterisk-Users] Re: How is Teliax ?   asterisk at anime.net
• [Asterisk-Users] Re: How is Teliax ?   asterisk at anime.net
• [Asterisk-Users] Hinting   asterisk at anime.net
• [Asterisk-Users] polycom overlap dialing?   C F
• [Asterisk-Users] Hinting   James Texter
• [Asterisk-Users] Interrupting a call   Jorge Mendoza
• [Asterisk-Users] morcdr v0.1 released   Darren Wiebe
• [Asterisk-Users] Hinting   asterisk at anime.net
• [Asterisk-Users] Callback auto dialing   Darren Wiebe
• [Asterisk-Users] Blocked channels, according to our telco... leading to CONGESTION status   Edwin Groothuis
• [Asterisk-Users] New Skype<>SIP gateway   Michael Graves
• [Asterisk-Users] Need More Simultaneous Voice Channel Capacity on Asterisk   Tadepalli, Hari K
• [Asterisk-Users] RE: Need More Simultaneous Voice Channel Capacity on Asterisk   Tadepalli, Hari K
• [Asterisk-Users] RE: Need More Simultaneous Voice Channel Capacity on Asterisk   Tadepalli, Hari K
• [Asterisk-Users] Dial cmd has delay for the last dialed number on FXO   Mojo with Horan & Company, LLC
• [Asterisk-Users] Re: Random music not so 'random'   Roger McCoy
• [Asterisk-Users] random beeps during calls   Sean Garland
• [Asterisk-Users] Blocked channels, according to our telco... leading to CONGESTION status   C F
• [Asterisk-Users] Monitor or mixmonitor   Wai Wu
• [Asterisk-Users] GoDaddy royally screws over aussievoip.com.au and soft-swtich.org   Rob Thomas
• [Asterisk-Users] Interrupting a call   Eric Jacksch
• [Asterisk-Users] MeetMe/Asterisk Timer   Kelvin Williams
• [Asterisk-Users] Building Asterisk embedded device   Brian Capouch
• [Asterisk-Users] R2 protocol error   Dennis Nacino
• [Asterisk-Users] DID billing   Darren Wiebe
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Kevin P. Fleming
• [Asterisk-Users] RE: Re: Re: Compatible Asterisk Connectivity Cards :Sangoma   Tomislav Parčina
• [Asterisk-Users] Re: ooh323 and g729 - any issue?   Tomislav Parčina
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Olivier Krief
• [Asterisk-Users] Re: Junghanns and Digium TDM400?   Olivier Krief
• [Asterisk-Users] MeetMe/Asterisk Timer   Alexander Chemeris
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Olivier Krief
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Lee Archer
• [Asterisk-Users] QSIG and multi-PBX receptionnist   Olivier Krief
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   ali asma
• [Asterisk-Users] Asterisk, QSIG and Tenovis PBX?   Dinesh Nair
• [Asterisk-Users] Re: Junghanns and Digium TDM400?   Koopmann, Jan-Peter
• [Asterisk-Users] Dial(L(x...)) distinct to SET TIMEOUT(absolute)   Mike Daws
• [Asterisk-Users] E1 te110p problem   Antonio Almodóvar
• [Asterisk-Users] Fax over 2 bridged TE110P channels   Alessio Focardi
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   ali asma
• [Asterisk-Users] IAX: Auto-congesting call due to slow response   Mimmus
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   Lee Archer
• [Asterisk-Users] Auto Attendant Question   Lyndon Labuschagne
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   ali asma
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   Lee Archer
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   ali asma
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   ali asma
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   Lee Archer
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   Lee Archer
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   Michiel van Baak
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   ali asma
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   Lee Archer
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   ali asma
• [Asterisk-Users] SIP Responsecodes   Olle E Johansson
• [Asterisk-Users] H323 problems   FaberK
• [Asterisk-Users] New User's Query , which card TE411P or TE410P   John Joseph
• [Asterisk-Users] Re: Auto Attendant Question   Tomislav Parčina
• [Asterisk-Users] Re: Auto Attendant Question   Lyndon Labuschagne
• [Asterisk-Users] Re: H323 problems   Tomislav Parčina
• [Asterisk-Users] Re: How is Teliax ?   Rich Adamson
• [Asterisk-Users] Re: H323 problems   Dinesh Nair
• [Asterisk-Users] RE: Monitor or mixmonitor   Jonathan k. Creasy
• [Asterisk-Users] Hinting   Jonathan k. Creasy
• [Asterisk-Users] MeetMe/Asterisk Timer   Matt Florell
• [Asterisk-Users] chan-capi: Sending digits on a bri (isdn) d-channel   Raoul Bönisch
• [Asterisk-Users] Pickup() h323   Pavel Jezek
• [Asterisk-Users] Asterisk cmd ZapSendKeypadFacility   Raoul Bönisch
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Rich Adamson
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Rich Adamson
• [Asterisk-Users] E1 te110p problem   Toke
• [Asterisk-Users] chan-capi: Sending digits on a bri (isdn) d-channel   Armin Schindler
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   ali asma
• [Asterisk-Users] Re: How is Teliax ?   Rich Adamson
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Olivier Krief
• [Asterisk-Users] Hangupcause is not enough on PRI   Pibix
• [Asterisk-Users] Hangupcause is not enough on PRI   Peeramate at SIPPhone Thailand
• [Asterisk-Users] Hangupcause is not enough on PRI   Andrew Kohlsmith
• [Asterisk-Users] Blocked channels, according to our telco... leading to CONGESTION status   Andrew Kohlsmith
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Andrew Kohlsmith
• [Asterisk-Users] Hangupcause is not enough on PRI   Wai Wu
• [Asterisk-Users] IAX connection refused between 2 asterisks 1.2.5   Marco Mouta
• [Asterisk-Users] Jitter in SIP calls?   Michael Welter
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   Derek Whitten
• [Asterisk-Users] Need to Install Fax to Email feature   Rob McKrill
• [Asterisk-Users] MeetMe/Asterisk Timer   Derek Whitten
• [Asterisk-Users] Fax over 2 bridged TE110P channels   cguy at bigpond.net.au
• [Asterisk-Users] Frustrated with echo...   Steve Jones
• [Asterisk-Users] IAX: Auto-congesting call due to slow response   Pavel Jezek
• [Asterisk-Users] Realtime Database Lookup   Dan Journo
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   ali asma
• [Asterisk-Users] Possible PRI fault?   Lee Archer
• [Asterisk-Users] Phones are all auto answering   Christian Buchter
• [Asterisk-Users] Phones are all auto answering   C F
• [Asterisk-Users] Phones are all auto answering   Christian Buchter
• [Asterisk-Users] Possible PRI fault?   Andrew Kohlsmith
• [Asterisk-Users] Phones are all auto answering   Pete Barnwell
• [Asterisk-Users] Phones are all auto answering   Christian Buchter
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   Joshua Colp
• [Asterisk-Users] IAX connection refused between 2 asterisks 1.2.5   Joshua Colp
• [Asterisk-Users] Can't recieve Fax: No carrier detected - Asterisk + iaxmodem + Hylafax   Pimjai Wesnarat
• [Asterisk-Users] Possible PRI fault?   Lee Archer
• [Asterisk-Users] Can't recieve Fax: No carrier detected - Asterisk + iaxmodem + Hylafaxv --- sorry.wrong log.   Pimjai Wesnarat
• [Asterisk-Users] Monitor or mixmonitor   Gary Richardson
• [Asterisk-Users] RE: Need More Simultaneous Voice Channel Capacity on Asterisk   Matt Roth
• [Asterisk-Users] RE: Need More Simultaneous Voice Channel Capacity on Asterisk   Tadepalli, Hari K
• [Asterisk-Users] Need to Install Fax to Email feature   Wasif
• [Asterisk-Users] Possible PRI fault?   Andrew Kohlsmith
• [Asterisk-Users] Monitor or mixmonitor   Wai Wu
• [Asterisk-Users] X100P small test gives cracking sound at the voip side   Erick Perez
• [Asterisk-Users] Re: E1 te110p problem   Infobox Peru
• [Asterisk-Users] Re: Random music not so 'random'   Nathan Bowyer
• [Asterisk-Users] 1.2.6 doesn't use mpg123?   Matt Roth
• [Asterisk-Users] Any Aheeva Users?   Steve Totaro
• [Asterisk-Users] Can't recieve Fax: No carrier detected - Ast erisk + iaxmodem + Hylafaxv --- sorry.wrong log.   Colin Anderson
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Jim Hanlon
• [Asterisk-Users] Can't recieve Fax: No carrier detected - Asterisk + iaxmodem + Hylafaxv --- sorry.wrong log.   Alessio Focardi
• [Asterisk-Users] Confused on Agents and Queues   Alan Ferrency
• [Asterisk-Users] Confused on Agents and Queues   Matt
• [Asterisk-Users] voicemail context issue   Aaron Daniel
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   John Novack
• [Asterisk-Users] Ideal setup for PRI/T1 and TE110P   JT Zemp
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Andrew Kohlsmith
• [Asterisk-Users] Re: Random music not so 'random'   Noah Miller
• [Asterisk-Users] voicemail context issue   Dov Bigio
• [Asterisk-Users] Fax over 2 bridged TE110P channels   Remco Barende
• [Asterisk-Users] Can't get Pickup app working   Álvaro Palma
• [Asterisk-Users] Can't get Pickup app working   Joshua Colp
• [Asterisk-Users] IAX connection refused between 2 asterisks 1.2.5   Marco Mouta
• [Asterisk-Users] Need to Install Fax to Email feature   Jonathan Attwood
• [Asterisk-Users] IAX connection refused between 2 asterisks 1.2.5   Joshua Colp
• [Asterisk-Users] IAX connection refused between 2 asterisks 1.2.5   Noah Miller
• [Asterisk-Users] IAX connection refused between 2 asterisks 1.2.5   Marco Mouta
• [Asterisk-Users] Anyone have a definitive list of Manager events per category?   Josh McAllister
• [Asterisk-Users] IAX connection refused between 2 asterisks 1.2.5   Joshua Colp
• [Asterisk-Users] Ideal Setup for T1/PRI and TE110P - second try   JT Zemp
• [Asterisk-Users] Ideal setup for PRI/T1 and TE110P   Robert Woodcock
• [Asterisk-Users] Can't recieve Fax: No carrier detected - Asterisk + iaxmodem + Hylafaxv --- sorry.wrong log.   Anton Krall
• [Asterisk-Users] Anyone have a definitive list of Manager eventsper category?   Wai Wu
• [Asterisk-Users] queueue recording and what to do next   Anton Krall
• [Asterisk-Users] Can't recieve Fax: No carrier detected - Asterisk + iaxmodem + Hylafaxv --- sorry.wrong log.   Rusty Dekema
• [Asterisk-Users] Ideal Setup for T1/PRI and TE110P - second try   Don Pobanz
• [Asterisk-Users] Jitter in SIP connection   Michael Welter
• [Asterisk-Users] Anyone have a definitive list of Manager eventsper category?   Josh McAllister
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Michiel van Baak
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Andrew Kohlsmith
• [Asterisk-Users] Pickup() h323   Jeremy McNamara
• [Asterisk-Users] WebMeetme defines.php?   Jordan Novak
• [Asterisk-Users] queueue recording and what to do next   Michiel van Baak
• [OT]Re: [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Michiel van Baak
• [Asterisk-Users] Opensource solutions to SPIT   Andy Tan
• [Asterisk-Users] Anyone have a definitive list of Managereventsper category?   Wai Wu
• [Asterisk-Users] Applying patch.   Wai Wu
• [Asterisk-Users] AST eating CPU 100%->Resource temporarily unavailable   Oscar Carriles
• [Asterisk-Users] Any Aheeva Users?   Kevin P. Fleming
• [Asterisk-Users] voicemail context issue   Kevin P. Fleming
• [Asterisk-Users] Any Aheeva Users?   Wai Wu
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Paulo Scardine
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   MBIT Technologies
• [Asterisk-Users] Hangupcause is not enough on PRI   Mojo with Horan & Company, LLC
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Paulo Scardine
• [Asterisk-Users] Asterisk in production as a fax server, anyone?   Paulo Scardine
• [Asterisk-Users] not transmit audio on sipura 941   Michael Blood
• [Asterisk-Users] not transmit audio on sipura 941   C F
• [Asterisk-Users] R2 protocol error   Moises Silva
• [Asterisk-Users] queueue recording and what to do next   Anton Krall
• [Asterisk-Users] Distinctive Ring on SPA941   Cory Hawkless
• [Asterisk-Users] Distinctive Ring on SPA941   Hadley Rich
• [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?   Anton Krall
• [Asterisk-Users] Distinctive Ring on SPA941   Cory Hawkless
• [Asterisk-Users] Set(CDR(anything_but_userfield_or_accountcode)=bla) broken?   Eric Lyons
• [Asterisk-Users] Set(CDR(anything_but_userfield_or_accountcode)=bla) broken?   Kevin P. Fleming
• [Asterisk-Users] R2 protocol error   Oscar Carriles
• [Asterisk-Users] IAX connection refused between 2 asterisks 1.2.5   Marco Mouta
• [Asterisk-Users] Need 25-50 Linksys boxes   Andy
• [Asterisk-Users] R2 protocol error   Dennis Nacino
• [Asterisk-Users] Too many open files   Raymond Chen
• [Asterisk-Users] WebMeetme defines.php?   Richard OSS
• [Asterisk-Users] Asterisk in production as a fax server, anyone?   Julio Arruda
• [Asterisk-Users] speech rec what works   A_ Navone
• [Asterisk-Users] voipstunt: "Forbidden - wrong password ..."   Ronald Wiplinger
• [Asterisk-Users] speech rec what works   Kevin P. Fleming
• [Asterisk-Users] Pickup() h323   Hamid Hashemi
• [Asterisk-Users] Realtime Database Lookup   Kyle Sexton
• [Asterisk-Users] sip hang channels   Raymond Chen
• [Asterisk-Users] Frustrated with echo...   Lorentz Hinrichsen
• [Asterisk-Users] Frustrated with echo...   Kevin P. Fleming
• [Asterisk-Users] Frustrated with echo...   jason justman
• [Asterisk-Users] VoiceMail realtime not working in asterisk-1.2.6   asterisk user
• [Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting   Dinesh Nair
• [Asterisk-Users] Milliwatt Test Number List   William M. Sandiford
• [Asterisk-Users] OT: Polycom IP501 and Speed Dials - FIXED!   Avi Miller
• [Asterisk-Users] Pickup() h323   Jeremy McNamara
• [Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting   Avi Miller
• [Asterisk-Users] ASTCC: How to reset "in-use" flag automatically ?   JP Carballo
• [Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting   Dinesh Nair
• [Asterisk-Users] ASTCC: How to reset "in-use" flag automatically ?   Ronald Wiplinger
• [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Isaac Xiao
• [Asterisk-Users] Pickup() h323   Hamid Hashemi
• [Asterisk-Users] some problems with asterisk and E1   Sebastian Reitenbach
• [Asterisk-Users] Asterisk-addons compiling problem   René Enskat [Teamware GmbH]
• [Asterisk-Users] Asterisk svn starting problem   René Enskat [Teamware GmbH]
• [Asterisk-Users] Asterisk svn starting problem   Dave Cotton
• [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Matt Florell
• [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me!   ali asma
• AW: [Asterisk-Users] Asterisk svn starting problem   René Enskat [Teamware GmbH]
• [Asterisk-Users] extensions.conf - switch => statement?   Benoit Panizzon
• [Asterisk-Users] ASTCC: How to reset "in-use" flag automatically ?   JP Carballo
• [Asterisk-Users] VPB cannot call out   hensem boy
• [Asterisk-Users] Fax over 2 bridged TE110P channels   Alessio Focardi
• [Asterisk-Users] Hangup Supervision Issue on Digium TDM11B   sdcharly at gmail.com
• [Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting   Dinesh Nair
• [Asterisk-Users] Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)   Marco Mouta
• [Asterisk-Users] Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)   Pimjai Wesnarat
• [Asterisk-Users] Error Header field Via   Il Neofita
• [Asterisk-Users] Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)   Marco Mouta
• [Asterisk-Users] Re: E1 te110p problem   Antonio Almodóvar
• [Asterisk-Users] Dial Plan Logic Problem   Jon Farmer
• [Asterisk-Users] Got SIP response 302 "Moved temporarely"   Benoit Panizzon
• [Asterisk-Users] Dial Plan Logic Problem   Doug Lytle
• [Asterisk-Users] Dial Plan Logic Problem   CC Jay
• [Asterisk-Users] Frustrated with echo...   Lorentz Hinrichsen
• [Asterisk-Users] Dial Plan Logic Problem   Steve Davies
• [Asterisk-Users] Frustrated with echo...   Eric "ManxPower" Wieling
• [Asterisk-Users] ASTCC: How to reset "in-use" flag automatically ?   Ronald Wiplinger
• [Asterisk-Users] voipstunt: "Forbidden - wrong password ..."   Andrei Sotirov
• [Asterisk-Users] Querying number of people in a call queue from dialplan   Gareth Blades
• [Asterisk-Users] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)   Matt
• [Asterisk-Users] Frustrated with echo...   Steve Jones
• [Asterisk-Users] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)   Sig Lange
• [Asterisk-Users] Got SIP response 302 "Moved temporarely"   Olle E. Johansson
• [Asterisk-Users] oh323 - cant load module   yusuf
• [Asterisk-Users] Asterisk svn starting problem   BJ Weschke
• [Asterisk-Users] WebMeetme defines.php?   Mike Clark
• [Asterisk-Users] Frustrated with echo...   Andrew Kohlsmith
• [Asterisk-Users] Asterisk in production as a fax server, anyone?   Paulo Scardine
• [Asterisk-Users] Frustrated with echo...   Lorentz Hinrichsen
• [Asterisk-Users] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)   Ronald Wiplinger
• [Asterisk-Users] Frustrated with echo...   Rich Adamson
• [Asterisk-Users] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)   Matt
• [Asterisk-Users] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)   Steve Underwood
• [Asterisk-Users] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)   Matt
• [Asterisk-Users] ASTCC: How to reset "in-use" flag automatically ?   Ronald Wiplinger
• [Asterisk-Users] queue issue   Dov Bigio
• [Asterisk-Users] Frustrated with echo...   Steve Jones
• [Asterisk-Users] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)   Matt Florell
• [Asterisk-Users] Dial Plan Logic Problem   Jon Farmer
• [Asterisk-Users] SIP T   Jon Weisman
• [Asterisk-Users] How to check if a phone / line is used?   Paul Zimm
• [Asterisk-Users] New Skype<>SIP gateway   Shad Mortazavi
• [Asterisk-Users] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)   Matt
• [Asterisk-Users] chan_modem_i4l delay   Alain Degreffe
• [Asterisk-Users] New Skype<>SIP gateway   Derek Whitten
• [Asterisk-Users] How to check if a phone / line is used?   Colin Anderson
• [Asterisk-Users] long delay between "Ring Begin" and "SIP/XXX is ringing"   lartc
• [Asterisk-Users] Frustrated with echo...   Rich Adamson
• [Asterisk-Users] chan_modem_i4l delay   Rene Kluwen
• [Asterisk-Users] Problem with setting ringtones on Cisco 7960 phone.   Jeremy Koski
• [Asterisk-Users] VPB cannot call out   Dovid Bender
• [Asterisk-Users] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)   Matt Florell
• [Asterisk-Users] GoDaddy royally screws over aussievoip.com.au and soft-swtich.org   Dovid Bender
• [Asterisk-Users] Patch 5779 on 1.0.9?   Colin Anderson
• [Asterisk-Users] can't start chan_capi with asterisk group   amaury BOSSE
• [Asterisk-Users] long delay between "Ring Begin" and "SIP/XXX is ringing"   Rich Adamson
• [Asterisk-Users] chan_modem_i4l delay   Armin Schindler
• [Asterisk-Users] queue issue   Dinesh Nair
• [Asterisk-Users] SIP Asterisk Polycom Reinvite   Damon Estep
• [Asterisk-Users] fax server functionality on Asterisk   Frank Ochmann
• [Asterisk-Users] RE: Milliwatt Test Number List   William M. Sandiford
• [Asterisk-Users] Problem with setting ringtones on Cisco 7960 phone.   Rich Adamson
• [Asterisk-Users] Master.csv Shell Script   Jeremy
• [Asterisk-Users] TE110P errors   Kenneth Lussier
• [Asterisk-Users] chan_modem_i4l delay   info at lolisoft.be
• [Asterisk-Users] GoDaddy royally screws over aussievoip.com.au and soft-swtich.org   Ronald Wiplinger
• [Asterisk-Users] Need 25-50 Linksys boxes   Noah Miller
• [Asterisk-Users] SHOWCHANINFO Not Working   Dan Journo
• [Asterisk-Users] fax server functionality on Asterisk   Derek Whitten
• [Asterisk-Users] Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)   Noah Miller
• [Asterisk-Users] fax server functionality on Asterisk   Bob McDowell
• [Asterisk-Users] Re: Asterisk start/stop   Steven
• [Asterisk-Users] GoDaddy royally screws over aussievoip.com.au and soft-swtich.org   Dan Journo
• [Asterisk-Users] RE: Milliwatt Test Number List   Olivier Krief
• [Asterisk-Users] chan_modem_i4l delay   info at lolisoft.be
• [Asterisk-Users] Favorite softphone with command line interface   Olivier Krief
• [Asterisk-Users] Master.csv Shell Script   Mojo with Horan & Company, LLC
• [Asterisk-Users] Fax over 2 bridged TE110P channels   Olivier Krief
• [Asterisk-Users] Asterisk support to Tornado M5 IP Phones   Juan Carlos Huerta
• [Asterisk-Users] legacy Alcatel 4200/4400 and Asterisk (QSIG/PRI) and callerid   Miroslav HOSTINSKY
• [Asterisk-Users] long delay between "Ring Begin" and "SIP/XXX is ringing"   lartc
• [Asterisk-Users] Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)   Marco Mouta
• [Asterisk-Users] one-waysilence during calls   Tommaso Calosi
• [Asterisk-Users] SIP client looses register and then i need to restart my pc to get registered on Asterisk 1.2.5   Marco Mouta
• [Asterisk-Users] can't start chan_capi with asterisk group   Armin Schindler
• [Asterisk-Users] Problem with setting ringtones on Cisco 7960 phone.   Jeremy Koski
• [Asterisk-Users] legacy Alcatel 4200/4400 and Asterisk (QSIG/PRI) and callerid   Krzysztof Drewicz
• [Asterisk-Users] Asterisk svn starting problem   Olle E Johansson
• [Asterisk-Users] Asterisk on BSD?   Bruce Ferrell
• [Asterisk-Users] fax server functionality on Asterisk   Paulo Scardine
• [Asterisk-Users] zaphfc NT Mode. Extension not recognized...   Benoit Panizzon
• [Asterisk-Users] Asterisk RealTime queue - periodic-announce   Kristian Marcroft
• [Asterisk-Users] Asterisk on BSD?   Michiel van Baak
• [Asterisk-Users] The Asterisk bug tracker :: please think twice before opening a report!   Olle E Johansson
• [Asterisk-Users] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)   Matt
• [Asterisk-Users] Queues - Dumb question   Franklin Webb
• [Asterisk-Users] transforming g729 files to wav files   Tofik Suleymanov
• [Asterisk-Users] SIP T   Olle E Johansson
• [Asterisk-Users] SIP T   Jon Weisman
• [Asterisk-Users] Problem with setting ringtones on Cisco 7960 phone.   Pavel Jezek
• [Asterisk-Users] IAX2 Origination Problem   CFN - Jan Serve
• [Asterisk-Users] IAX2 Origination Problem   Joshua Colp
• [Asterisk-Users] transforming g729 files to wav files   Noah Miller
• [Asterisk-Users] RE: Problem with setting ringtones on Cisco 7960 phone.   Paul A. Pringle
• [Asterisk-Users] Sending Access codes to a 5EE switch.   Gary Ritter
• [Asterisk-Users] IAX2 Origination Problem   CFN - Jan Serve
• [Asterisk-Users] Pickup() h323   Dan Austin
• [Asterisk-Users] Sending Access codes to a 5EE switch.   Jon Weisman
• [Asterisk-Users] RE: Problem with setting ringtones on Cisco 7960 phone.   Rich Adamson
• [Asterisk-Users] Sending Access codes to a 5EE switch.   Andrew Kohlsmith
• [Asterisk-Users] transforming g729 files to wav files   Darrell Long
• [Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting   Avi Miller
• [Asterisk-Users] How to restrict simultaneous phone registrations   Bryan Mahin
• [Asterisk-Users] What does this error mean "app.c: Huh....? no dial for indications?"   Chuck Bunn
• [Asterisk-Users] What causes deadlock?   Chuck Bunn
• [Asterisk-Users] cisco 7960   Jimmy Smith
• [Asterisk-Users] WebMeetme Problem Please help!!!   Jordan Novak
• [Asterisk-Users] Fedora Core 4 - problem with kernel 2.6.16-1.2069_FC4   William M Conlon
• [Asterisk-Users] How to restrict simultaneous phone registrations   Eric "ManxPower" Wieling
• [Asterisk-Users] Sending Access codes to a 5EE switch.   Eric "ManxPower" Wieling
• [Asterisk-Users] WebMeetme Problem Please help!!!   Dan Austin
• [Asterisk-Users] cisco 7960   Greg Oliver
• [Asterisk-Users] cisco 7960   Aaron Daniel
• [Asterisk-Users] ASTCC: How to reset "in-use" flag automatically ?   JP Carballo
• [Asterisk-Users] How to restrict simultaneous phone registrations   Ronald Wiplinger
• [Asterisk-Users] Setting ptime attribute in SDP invite   Eric Bishop
• [Asterisk-Users] Running into problems with the Digital Receptionist (Callers are not redirected to it)-   Maxx Lobo
• [Asterisk-Users] Running into problems with the Digital Receptionist (Callers are not redirected to it)-   Maxx Lobo
• [Asterisk-Users] WOW! Sphinx is awesome... but....(asterisk+sphinx+menus)   Quark IT - Hilton Travis
• [Asterisk-Users] Running into problems with the Digital Receptionist (Callers are not redirected to it)-   Maxx Lobo
• [Asterisk-Users] How to restrict simultaneous phone registrations   Bryan Mahin
• [Asterisk-Users] How to restrict simultaneous phone registrations   Bill Gibbs
• [Asterisk-Users] How to restrict simultaneous phone registrations   Bryan Mahin
• [Asterisk-Users] E-911 Canada Info - Hot Off the Press   Bob's Leaky News Service
• [Asterisk-Users] How to restrict simultaneous phone registrations   Derek Whitten
• [Asterisk-Users] Sending Access codes to a 5EE switch.   Jon Weisman
• [Asterisk-Users] How to restrict simultaneous phone registrations   Darren Wiebe
• [Asterisk-Users] How to restrict simultaneous phone registrations   Alexander Lopez
• [Asterisk-Users] Fedora Core 4 - problem with kernel2.6.16-1.2069_FC4   Michael L. Young
• [Asterisk-Users] NTT (Japan) Voip Adapter   Jason Frisch
• [Asterisk-Users] How to restrict simultaneous phone registrations   Bryan Mahin
• [Asterisk-Users] Display Name   Alexander Lopez
• [Asterisk-Users] How to restrict simultaneous phone registrations   Bryan Mahin
• [Asterisk-Users] Display Name   Alexander Lopez
• [Asterisk-Users] What causes deadlock?   Andy Kuo
• [Asterisk-Users] asterisk redundancy   Tharanga
• [Asterisk-Users] Asterisk on BSD?   Sebastian Reitenbach
• [Asterisk-Users] Fwd: [dmuars] Eh up - March 144 results altered   Peter Bowyer
• [Asterisk-Users] Re: [dmuars] Eh up - March 144 results altered   Peter Bowyer
• [Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting   Dinesh Nair
• [Asterisk-Users] VPB cannot call out   hensem boy
• [Asterisk-Users] Chan-sccp - Asterisk dies   Tomislav Parčina
• [Asterisk-Users] CallerID   Miles Scruggs
• [Asterisk-Users] How to restrict simultaneous phone registrations   Michiel van Baak
• [Asterisk-Users] ASTCC: How to reset "in-use" flag automatically ?   Ronald Wiplinger
• [Asterisk-Users] IAX: Auto-congesting call due to slow response   Mimmus
• [Asterisk-Users] fax server functionality on Asterisk   Mimmus
• [Asterisk-Users] legacy Alcatel 4200/4400 and Asterisk (QSIG/PRI)and callerid   Mimmus
• [Asterisk-Users] chan_modem_i4l delay again..   Alain Degreffe
• [Asterisk-Users] Re: Re: H323 problems   Tomislav Parčina
• [Asterisk-Users] legacy Alcatel 4200/4400 and Asterisk (QSIG/PRI)and callerid   Krzysztof Drewicz
• [Asterisk-Users] legacy Alcatel 4200/4400 andAsterisk (QSIG/PRI)and callerid   Mimmus
• [Asterisk-Users] Hinting a conference room   Alessio Focardi
• [Asterisk-Users] Re: Hangupcause is not enough on PRI   Tomislav Parčina
• [Asterisk-Users] IVR : Can't hear my message   Antoine LOUIS
• [Asterisk-Users] Fwd: Hangup Supervision   sdcharly at gmail.com
• [Asterisk-Users] Incoming call redirected to mobile   Julian Lyndon-Smith
• [Asterisk-Users] How to restrict simultaneous phone registrations   Eric "ManxPower" Wieling
• [Asterisk-Users] Call transfer to cell phone   Giuseppe
• [Asterisk-Users] Incoming call redirected to mobile   Eric "ManxPower" Wieling
• [Asterisk-Users] Dial out on Zap   Pimjai Wesnarat
• [Asterisk-Users] Re: Can't get Pickup app working   Tomislav Parčina
• AW: [Asterisk-Users] Dial out on Zap   Marcus.Rothe at bertelsmann.de
• [Asterisk-Users] Call transfer to cell phone [UPDATE]   Giuseppe
• [Asterisk-Users] can't start chan_capi with asterisk group   amaury BOSSE
• [Asterisk-Users] not get ring tone with chan-capi and avm b1   Ricardo
• [Asterisk-Users] can't start chan_capi with asterisk group   Armin Schindler
• [Asterisk-Users] Re: queue issue   Tomislav Parčina
• [Asterisk-Users] qozap errors on junghanns QuadBRI   Andrzej Wolski
• [Asterisk-Users] FXO/FXS and E1 in same system   yusuf
• [Asterisk-Users] How to restrict simultaneous phone registrations   Jonathan k. Creasy
• [Asterisk-Users] qozap errors on junghanns QuadBRI   Krzysztof Drewicz
• [Asterisk-Users] Asterisk dialing over asterisk to PSTN   René Enskat [Teamware GmbH]
• [Asterisk-Users] FXS module failed   Gabriel Perez S.
• [Asterisk-Users] Re: What causes deadlock?   Tomislav Parčina
• [Asterisk-Users] Re: What causes deadlock?   Raymond Chen
• [Asterisk-Users] Re: CallerID   Tomislav Parčina
• [Asterisk-Users] Questions on call recording and conference.   Dinesh Nair
• [Asterisk-Users] TDM2400P problems   Sean Cook
• [Asterisk-Users] TDM2400P problems   Tim Jackson
• [Asterisk-Users] Voicemaster   Benni A. Aswin
• [Asterisk-Users] CallerID   Waldo Rubinstein
• [Asterisk-Users] TDM2400P problems   Time Bandit
• [Asterisk-Users] Re: queue issue   Lenz
• [Asterisk-Users] Fedora Core 4 - problem with kernel 2.6.16-1.2069_FC4   Bob McDowell
• [Asterisk-Users] Increase volume on trunk   Sam Tam
• [Asterisk-Users] Using Call Progress   Eric Buruschkin
• [Asterisk-Users] (no subject)   Marco Maiolini
• [Asterisk-Users] Call Parking and multiple contexts   Waldo Rubinstein
• [Asterisk-Users] Re: queue issue   Dinesh Nair
• [Asterisk-Users] Open channels   Tomislav Parčina
• [Asterisk-Users] Hinting a conference room   Alexander Lopez
• [Asterisk-Users] Using Call Progress   Kerry Garrison
• [Asterisk-Users] FXO/FXS and E1 in same system   Ondrej Valousek
• [Asterisk-Users] Re: What causes deadlock?   From PH
• [Asterisk-Users] Call Parking and multiple contexts   Alexander Lopez
• [Asterisk-Users] audiocodes with asterisk:- newbie   vivek at staff.ownmail.com
• [Asterisk-Users] pause / unpausequeuemember   Dov Bigio
• [Asterisk-Users] chan_sccp and hinting   Aaron Daniel
• [Asterisk-Users] Call Parking and multiple contexts   Waldo Rubinstein
• [Asterisk-Users] Routing SIP calls via URI   Joao Pereira
• [Asterisk-Users] Using Call Progress   Rich Adamson
• [Asterisk-Users] Pickup() h323   Dinesh Nair
• [Asterisk-Users] Increase volume on trunk   Rich Adamson
• [Asterisk-Users] Cisco 7960 - hints   Sean Cook
• [Asterisk-Users] Asterisk behind NAT   Joao Pereira
• [Asterisk-Users] Planet VIP-320 DECT gateway with Asterisk?   Louis-David Mitterrand
• [Asterisk-Users] Cisco 7960 - hints   Aaron Daniel
• [Asterisk-Users] How to restrict simultaneous phone registrations   Darren Wiebe
• [Asterisk-Users] Cisco 7960 - hints   Sean Cook
• [Asterisk-Users] Asterisk behind NAT   Kerry Garrison
• [Asterisk-Users] chan_sccp and hinting   Michiel van Baak
• [Asterisk-Users] New user needs a hand starting   Steve
• [Asterisk-Users] Soporte   Will Velez
• [Asterisk-Users] Pickup() h323   Pavel Jezek
• [Asterisk-Users] 7970 SIP   Armand Fumal
• [Asterisk-Users] Cisco 7960 - hints   Aaron Daniel
• [Asterisk-Users] increasing volume level to console/dsp   Jerry Geis
• [Asterisk-Users] Asterisk behind NAT   Giridhar Reddy Bandi
• [Asterisk-Users] Networld Interop, Vegas 2006   Daniel Hazelbaker
• [Asterisk-Users] VoiceMail realtime not working in asterisk-1.2.6   Nathan Bowyer
• [Asterisk-Users] Asterisk behind NAT   Joao Pereira
• [Asterisk-Users] Asterisk behind NAT   hgaillac-sip at yahoo.fr
• [Asterisk-Users] How to restrict simultaneous phone registrations   Bryan Mahin
• [Asterisk-Users] chan_sccp and hinting   Aaron Daniel
• [Asterisk-Users] Voicemaster   Benni A. Aswin
• [Asterisk-Users] How to restrict simultaneous phone registrations   Bryan Mahin
• [Asterisk-Users] How to restrict simultaneous phone registrations   Bryan Mahin
• [Asterisk-Users] How to restrict simultaneous phone registrations   Bryan Mahin
• [Asterisk-Users] Auto Attendant Question   Dovid Bender
• [Asterisk-Users] Stupid newbie question   Ira
• [Asterisk-Users] Hinting a conference room   Alessio Focardi
• [Asterisk-Users] queueue recording and what to do next   Matt Roth
• [Asterisk-Users] Asterisk behind NAT   Andre Ruiz
• [Asterisk-Users] Too many open files   Matt Roth
• [Asterisk-Users] Sending Access codes to a 5EE switch.   Gary Ritter
• [Asterisk-Users] Networld Interop, Vegas 2006   Kevin P. Fleming
• [Asterisk-Users] TDM400P and Junghans quadBRI   Steve Kennedy
• [Asterisk-Users] Pickup() h323   Dan Austin
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Dan Austin
• [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Matt Roth
• [Asterisk-Users] Re: What causes deadlock?   Matt
• [Asterisk-Users] chan_sccp and hinting   Michiel van Baak
• [Asterisk-Users] How to restrict simultaneous phone registrations   Derek Whitten
• [Asterisk-Users] chan_sccp and hinting   Ryan Amos
• [Asterisk-Users] ringing indication in handset when 2 extensions answer simultaneously?   asterisk at anime.net
• [Asterisk-Users] Auto Attendant Question   Matt
• [Asterisk-Users] Call/Contact Center.   Krzysztof Drewicz
• [Asterisk-Users] increasing volume level to console/dsp   Kerry Garrison
• [Asterisk-Users] Directory Issue   Darren Ellis
• [Asterisk-Users] Re: MWAnalyze question   Roger Schreiter
• [Asterisk-Users] Asterisk behind NAT   Giridhar Reddy Bandi
• [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma   Andrew Kohlsmith
• [Asterisk-Users] SIP Asterisk Polycom Reinvite   Matthew T. O'Connor
• [Asterisk-Users] Pickup() h323   Jeremy McNamara
• [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Matt Florell
• [Asterisk-Users] Call/Contact Center.   Matt Florell
• [Asterisk-Users] SIP Asterisk Polycom Reinvite   Damon Estep
• [Asterisk-Users] SIP Asterisk Polycom Reinvite   Sean Cook
• [Asterisk-Users] Pickup() h323   Dan Austin
• [Asterisk-Users] Call/Contact Center.   Steve Totaro
• [Asterisk-Users] Asterisk in FreeBSD   Mario Beltran
• [Asterisk-Users] Asterisk in FreeBSD   Joe Greco
• [Asterisk-Users] Pickup() h323   Jeremy McNamara
• [Asterisk-Users] chan_sccp and hinting   Aaron Daniel
• [Asterisk-Users] Call/Contact Center.   Erick Perez
• [Asterisk-Users] Originate   Daniel Laursen
• [Asterisk-Users] Directory Issue   Alexander Lopez
• [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long]   Colin Anderson
• [Asterisk-Users] Steps to make trunked iax2   jonny hashem
• [Asterisk-Users] queue/agent and macros?   Shaun
• [Asterisk-Users] digium card for xseries 346   Mark Quitoriano
• [Asterisk-Users] Originate   Ronald Wiplinger
• [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metres away with Cat 3 or telco wire [long]   James Harper
• [Asterisk-Users] Telasip   Jeremy
• [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long]   David Antognini
• [Asterisk-Users] Asterisk in FreeBSD   Michiel van Baak
• [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 m etres away with Cat 3 or telco wire [long]   Colin Anderson
• [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 m etre s away with Cat 3 or telco wire [long]   Colin Anderson
• [Asterisk-Users] What Media Gateway (connected via SS7) do you use   Heidi Mendoza
• [Asterisk-Users] Asterisk in FreeBSD   Miguel
• [Asterisk-Users] Suggested MeetMe feature: 'i' without review.   Jon Webster
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 21, Issue 38   Paul Davidson
• [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metres away with Cat 3 or telco wire [long]   James Harper
• [Asterisk-Users] Re: Fedora Core 4 - problem with kernel 2.6.16-1.2069_FC4   Axel Thimm
• [Asterisk-Users] Telasip   Tom Vile
• [Asterisk-Users] Telasip   Jeremy
• [Asterisk-Users] Telasip   Tom Vile
• [Asterisk-Users] Telasip   wendell hamilton
• [Asterisk-Users] SIP to another PBX w/ forwarding set   Greg Oliver
• [Asterisk-Users] # IP601's with POE per Catalyst 3560G-48PS   Jay Wilton
• [Asterisk-Users] # IP601's with POE per Catalyst 3560G-48PS   Greg Oliver
• [Asterisk-Users] # IP601's with POE per Catalyst 3560G-48PS   Rich Adamson
• [Asterisk-Users] Problem with playing old gsm files   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] asterisk box as a voip gateway   Mark Quitoriano
• [Asterisk-Users] Integrics ITSP 1.6 released   Alistair Cunningham
• [Asterisk-Users] Integrics ITSP 1.6 released   Tom Vile
• [Asterisk-Users] Pickup Group in VPB   hensem boy
• [Asterisk-Users] # IP601's with POE per Catalyst 3560G-48PS   Jay Wilton
• [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 m etre s away with Cat 3 or telco wire [long]   David Antognini
• [Asterisk-Users] Steps to make trunked iax2   Andre Ruiz
• [Asterisk-Users] Cisco 7970 SIP   Armand Fumal
• [Asterisk-Users] Call/Contact Center.   Krzysztof Drewicz
• [Asterisk-Users] Look What 911 Will Cost in Canada   Bob's Leaky News Service
• [Asterisk-Users] Bell Canada Requests $987.14 Rate increase 911 / VOIP Providers   Bob's Leaky News Service
• [Asterisk-Users] Pickup() h323   Hamid Hashemi
• [Asterisk-Users] VoiceMail realtime not working in asterisk-1.2.6   asterisk user
• [Asterisk-Users] Call/Contact Center.   Matt Florell
• [Asterisk-Users] 0-ECRS   Nick Hoffman
• [Asterisk-Users] zaphfc NT Mode. Extension not recognized...   Benoit Panizzon
• [Asterisk-Users] Line in use   Miles Scruggs
• [Asterisk-Users] CRTC Stops Bell Canada 911 rate increase for 45 days only   Bob's Leaky News Service
• [Asterisk-Users] Dial Plan Problem with extensions ringing multiple phones connected on different * servers   Benoit Panizzon
• [Asterisk-Users] CallerID   Miles Scruggs
• [Asterisk-Users] IAX: Auto-congesting call due to slow response   Mimmus
• [Asterisk-Users] CallerID   Alejandro Vargas
• [Asterisk-Users] gotoif   Shaun
• [Asterisk-Users] Re: gotoif   Shaun
• [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long]   Ronan Mullally
• [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long]   Ronan Mullally
• [Asterisk-Users] Re: Suggested MeetMe feature: 'i' without review.   Tony Mountifield
• [Asterisk-Users] MINNESOTA: TwinCities Asterisk Users Group - Saturday 04/08/2006   Shane Young
• [Asterisk-Users] match callerid against outgoing calls   James Andrewartha
• [Asterisk-Users] gotoif   Doug Lytle
• [Asterisk-Users] 407 proxy authentication   hgaillac-sip at yahoo.fr
• [Asterisk-Users] queue/agent and macros?   Gareth Blades
• [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long]   BJ Weschke
• [Asterisk-Users] IVR : Can't hear my message   Dovid Bender
• [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long]   Andrew Kohlsmith
• [Asterisk-Users] suggestions on an IP T1 to TDM T1 gateway solution   Damon Estep
• [Asterisk-Users] Inbound PRI calls drop after 5 seconds using Sangoma A101   Joe Prosser
• [Asterisk-Users] Inbound PRI calls drop after 5 seconds using   Doug Lytle
• [Asterisk-Users] digium card for xseries 346   Kevin P. Fleming
• [Asterisk-Users] Inbound PRI calls drop after 5 seconds using   Wai Wu
• [Asterisk-Users] OT: local calling guide   Jonathan k. Creasy
• [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 m etre s away with Cat 3 or telco wire [long]   Colin Anderson
• [Asterisk-Users] queue/agent and macros?   Johann
• [Asterisk-Users] regexp in gotoif   Christian B
• [Asterisk-Users] CallerID   Colin Anderson
• [Asterisk-Users] Re: IVR: Cant hear my message   Dovid Bender
• [Asterisk-Users] OT: local calling guide   Joseph Tanner
• [Asterisk-Users] VPB cannot call out   Dovid Bender
• [Asterisk-Users] Uplink Skype2Sip   Giordano Grandis
• [Asterisk-Users] transfer call after advise   nik600
• [Asterisk-Users] editing the asterisk -addons makefile   Jordan Novak
• [Asterisk-Users] Fedora 'service asterisk start' problems   Bob McDowell
• [Asterisk-Users] Cisco 7912 Phones & XML   Adam Mattina
• [Asterisk-Users] Beeps and noises during calls   Sean Garland
• [Asterisk-Users] Look What 911 Will Cost in Canada   John Novack
• [Asterisk-Users] Beeps and noises during calls   Darrell Long
• [Asterisk-Users] Cisco 7912 Phones & XML   Ryan Amos
• [Asterisk-Users] Beeps and noises during calls   Zoa
• [Asterisk-Users] wellgate registration 3802   Jerry Geis
• [Asterisk-Users] Beeps and noises during calls   Zoa
• [Asterisk-Users] transfer call after advise   Christian B
• [Asterisk-Users] Bell Canada Requests $987.14 Rate increase 911 /VOIP Providers   mustardman29
• [Asterisk-Users] Attended Transfer howto   Miles Scruggs
• [Asterisk-Users] transforming g729 files to wav files   Tofik Suleymanov
• [Asterisk-Users] transforming g729 files to wav files   Tofik Suleymanov
• [Asterisk-Users] Bell Canada Requests $987.14 Rate increase 9 11 /VOIP Providers   Colin Anderson
• [Asterisk-Users] suggestions on an IP T1 to TDM T1 gateway solution   Damon Estep
• [Asterisk-Users] Telephony newbie need advice for integration Nortel MICS 4.1 with Asterisk via T1/E1 interface   CG
• [Asterisk-Users] ASTCC: How to reset "in-use" flag automatically ?   JP Carballo
• [Asterisk-Users] asterisk box as a voip gateway   JP Carballo
• [Asterisk-Users] CallerID   Technical Support
• [Asterisk-Users] Asterisk compiling problems...   Paul Hewlett
• [Asterisk-Users] asterisk box as a voip gateway   Infobox Peru
• [Asterisk-Users] Re: Fedora Core 4 - problem with kernel 2.6.16-1.2069_FC4   William M Conlon
• [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers   Douglas Garstang
• [Asterisk-Users] Soporte   jat at unixtrends.com
• [Asterisk-Users] Telephony newbie need advice for integration Nortel MICS 4.1 with Asterisk via T1/E1 interface   Bob McDowell
• [Asterisk-Users] Soporte   Bob McDowell
• [Asterisk-Users] Sending Access codes to a 5EE switch.   Michael Collins
• [Asterisk-Users] Audiconferencing System fon Asterisk   Erick Perez
• [Asterisk-Users] Uplink Skype2Sip   Erick Perez
• [Asterisk-Users] Beeps and noises during calls   Sean Garland
• Fwd: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Erick Perez
• [Asterisk-Users] Re: Attended Transfer howto   Shaun Reitan
• [Asterisk-Users] Re: gotoif   Shaun Reitan
• [Asterisk-Users] Re: queue/agent and macros?   Shaun Reitan
• [Asterisk-Users] Re: queue/agent and macros?   Shaun Reitan
• [Asterisk-Users] Beeps and noises during calls   Andrew Kohlsmith
• [Asterisk-Users] Sending Access codes to a 5EE switch.   Andrew Kohlsmith
• [Asterisk-Users] Call tracking through chan_agent using the Manager API   Paul Robins
• [Asterisk-Users] Beeps and noises during calls   Sean Garland
• [Asterisk-Users] Beeps and noises during calls   Guido Hecken
• [Asterisk-Users] can we lend a hand?   support at sjobeck.com
• [Asterisk-Users] simple wav ringtones?   Dr. Michael J. Chudobiak
• [Asterisk-Users] Fedora 'service asterisk start' problems   Matt
• [Asterisk-Users] can we lend a hand?   Tim Panton
• [Asterisk-Users] can we lend a hand?   TC
• [Asterisk-Users] Re: Fedora Core 4 - problem with kernel 2.6.16-1.2069_FC4   Axel Thimm
• [Asterisk-Users] Steps to make trunked iax2   jonny hashem
• [Asterisk-Users] Steps to make trunked iax2   Rich Adamson
• [Asterisk-Users] Re: gotoif   Jeremy Wadhams
• [Asterisk-Users] Bell Canada Requests $987.14 Rate increase 9 11 /VOIP Providers   Bob's Leaky News Service
• [Asterisk-Users] Look What 911 Will Cost in Canada   Bob's Leaky News Service
• [Asterisk-Users] Canada Nomadic 911 - From the Yes it will Screw Your Biz Dept   Bob's Leaky News Service
• [Asterisk-Users] Attended Transfer howto   Melcon Moraes
• [Asterisk-Users] Look What 911 Will Cost in Canada   John Novack (port)
• [Asterisk-Users] RE: can we lend a hand?   JR Richardson
• [Asterisk-Users] Beeps and noises during calls   mustardman29
• [Asterisk-Users] asterisk box as a voip gateway   Mark Quitoriano
• [Asterisk-Users] [OT] Centrex Question   Brian Capouch
• [Asterisk-Users] Announcing Astmanproxy 1.20   dave at popvox.com
• [Asterisk-Users] [OT] Centrex Question   Gabriel Afana
• [Asterisk-Users] [OT] Centrex Question   Gabriel Afana
• [Asterisk-Users] [OT] Centrex Question   Alexander Lopez
• [Asterisk-Users] [OT] Centrex Question   Brian Capouch
• [Asterisk-Users] [OT] Centrex Question   Tom Rymes
• [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers   Alexander Lopez
• [Asterisk-Users] Problems with registering iaxy   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] CallerID   Jay Milk
• [Asterisk-Users] May be OT , but comparing   ram
• [Asterisk-Users] Re: gotoif   Shaun
• [Asterisk-Users] Pickup() h323   Jeremy McNamara
• [Asterisk-Users] AAstra 9133i register double account.. ??   nik600
• [Asterisk-Users] Quintum ASM400 FXO configuration   Abdul Lateef
• [Asterisk-Users] [OT] Centrex Question   Rich Adamson
• [Asterisk-Users] HELP !!!!!   hgaillac-sip at yahoo.fr
• [Asterisk-Users] HELP !!!!!   Jeremy McNamara
• [Asterisk-Users] HELP !!!!!   hgaillac-sip at yahoo.fr
• [Asterisk-Users] HELP !!!!!   Jeremy McNamara
• [Asterisk-Users] HELP !!!!!   hgaillac-sip at yahoo.fr
• [Asterisk-Users] HELP !!!!!   hgaillac-sip at yahoo.fr
• FW: [Asterisk-Users] CallerID   Technical Support
• [Asterisk-Users] Problems getting Asterisk to connect to sipgate - Times out   Paul A Brown
• [Asterisk-Users] question about DISA   Ronaldo Chan
• [Asterisk-Users] Re: gotoif   Doug Lytle
• [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers   Moises Silva
• [Asterisk-Users] 407 proxy authentication   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Re: [OT] Centrex Question   Levolorman at aol.com
• [Asterisk-Users] Call/Contact Center.   Krzysztof Drewicz
• [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 m etre s away with Cat 3 or telco wire [long]   Colin Anderson
• [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers   Alexander Lopez
• [Asterisk-Users] Call parking query   Alex Brett
• [Asterisk-Users] quadBRI PCI ISDN on Suse Linux 10   Colin MacMillan
• [Asterisk-Users] queueue recording and what to do next   Anton Krall
• [Asterisk-Users] queueue recording and what to do next   Kevin P. Fleming
• [Asterisk-Users] unable to enable stutter dialtone   rkb at i.frys.com
• [Asterisk-Users] G723 -> ulaw codec problem   Sam Tam
• [Asterisk-Users] G723 -> ulaw codec problem   Kevin P. Fleming
• [Asterisk-Users] queueue recording and what to do next   Steve Totaro
• [Asterisk-Users] Re: [asterisk-dev] bug or bad chan_sip.c   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Re: [asterisk-dev] bug or bad chan_sip.c   hgaillac-sip at yahoo.fr
• [Asterisk-Users] ANI on a PRI   Steve Totaro
• [Asterisk-Users] unable to enable stutter dialtone   Eric "ManxPower" Wieling
• [Asterisk-Users] Voicemail 0 for operator call routing   broadbandvoice at comcast.net
• [Asterisk-Users] Force codec   Michael Strelnikov
• [Asterisk-Users] Force codec   Steve Totaro
• [Asterisk-Users] Force codec   Eric "ManxPower" Wieling
• [Asterisk-Users] Force codec   Brian Capouch
• [Asterisk-Users] Force codec   Michael Strelnikov
• [Asterisk-Users] How to set busy   Miles Scruggs
• [Asterisk-Users] Problems with registering iaxy   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] MACRO_RESULT=ABORT   Shaun
• [Asterisk-Users] How to set busy   C F
• [Asterisk-Users] How to set busy   Benoit Panizzon
• [Asterisk-Users] oh323.conf problem   Tomislav Parčina
• [Asterisk-Users] How to set busy   Miles Scruggs
• [Asterisk-Users] How to set busy   Miles Scruggs
• [Asterisk-Users] How to set busy   Peter J Dean
• [Asterisk-Users] [SPAM] Re: [asterisk-dev] Announcing Astmanproxy 1.20   Dinesh Nair
• [Asterisk-Users] Re: [asterisk-dev] Announcing Astmanproxy 1.20   Dinesh Nair
• [Asterisk-Users] how to communicate two PCs on LAN with Asterisk   KhojaS
• [Asterisk-Users] Disable 407 proxy authentication for outbound domains   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Re: [asterisk-dev] Disable 407 proxy authentication for outbound domains   Michiel van Baak
• [Asterisk-Users] Problems with registering iaxy   Tim Panton
• [Asterisk-Users] How to set busy   Thomas Winter
• [Asterisk-Users] How to set busy   Thomas Winter
• [Asterisk-Users] Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why?   Marco Mouta
• [Asterisk-Users] quadBRI PCI ISDN on Suse Linux 10   Paul Hewlett
• [Asterisk-Users] how to communicate two PCs on LAN with Asterisk   Steve Totaro
• [Asterisk-Users] meetme   Steve Totaro
• [Asterisk-Users] meetme   Steve Totaro
• [Asterisk-Users] meetme   Miles Scruggs
• [Asterisk-Users] meetme   Miles Scruggs
• [Asterisk-Users] meetme   Miles Scruggs
• [Asterisk-Users] Best ATA for general residential deployment??   broadbandvoice at comcast.net
• [Asterisk-Users] How to set busy   Eric "ManxPower" Wieling
• [Asterisk-Users] GXP-2000 and Voicemail   Waldo Rubinstein
• [Asterisk-Users] Newbie question - sip.conf incoming contexts   Steve Gladden
• [Asterisk-Users] Newbie question - sip.conf incoming contexts   Steve Gladden
• [Asterisk-Users] meetme   Alexander Lopez
• [Asterisk-Users] meetme   Steve Totaro
• [Asterisk-Users] oh323.conf problem   Yusuf
• [Asterisk-Users] meetme   Steve Totaro
• [Asterisk-Users] Best ATA for general residential deployment??   The VoIP Connection
• [Asterisk-Users] meetme   Miles Scruggs
• [Asterisk-Users] meetme   Bill
• [Asterisk-Users] GXP-2000 and Voicemail   Harald Holzer
• [Asterisk-Users] txfax tiff file format   Mohammad Shokuie
• [Asterisk-Users] meetme   Miles Scruggs
• [Asterisk-Users] meetme   Miles Scruggs
• [Asterisk-Users] meetme   Miles Scruggs
• [Asterisk-Users] How to set busy   C F
• [Asterisk-Users] Psgw   kevin ling
• [Asterisk-Users] meetme   Miles Scruggs
• [Asterisk-Users] meetme   Steve Totaro
• [Asterisk-Users] CallerID   Jay Milk
• [Asterisk-Users] meetme   Steve Totaro
• [Asterisk-Users] CallerID   Steve Totaro
• R: [Asterisk-Users] Psgw   Giordano Grandis
• [Asterisk-Users] Psgw   Brian Capouch
• [Asterisk-Users] GXP-2000 and Voicemail   Waldo Rubinstein
• [Asterisk-Users] Force codec   Kerry Garrison
• [Asterisk-Users] CallerID   Miles Scruggs
• [Asterisk-Users] GXP-2000 and Voicemail   Tim Litwiller
• [Asterisk-Users] GXP-2000 and Voicemail   Waldo Rubinstein
• [Asterisk-Users] Force codec   Brian Capouch
• [Asterisk-Users] 407 proxy authentication   JOAO CARLOS MOURA
• [Asterisk-Users] meetme   Miles Scruggs
• [Asterisk-Users] How to avoid "Avoiding deadlock..."   Joe
• [Asterisk-Users] meetme   Alexander Lopez
• [Asterisk-Users] meetme   Miles Scruggs
• [Asterisk-Users] Cisco 7960 problems   Paul A Brown
• [Asterisk-Users] Problem - Voicemail resets phone   Paul A Brown
• [Asterisk-Users] Cisco 7960 problems   Jon Farmer
• [Asterisk-Users] question about DISA   Tele Cost Price Reducer
• [Asterisk-Users] CallerID   Technical Support
• [Asterisk-Users] Cisco 7960 problems   Jeremy Wadhams
• [Asterisk-Users] Cisco 7960 problems   Paul A Brown
• [Asterisk-Users] Cisco 7960 problems   Paul A Brown
• [Asterisk-Users] Cisco 7960 problems   Jon Farmer
• [Asterisk-Users] Re: oh323.conf problem   Tomislav Parčina
• [Asterisk-Users] Force codec   pdhales at optusnet.com.au
• [Asterisk-Users] meetme   Steve Totaro
• [Asterisk-Users] ANI on a PRI   Steve Totaro
• [Asterisk-Users] Provisioning Server...   Andrew Stock
• [Asterisk-Users] MWI Problem   Eric Jacksch
• [Asterisk-Users] Force codec   Michael Strelnikov
• [Asterisk-Users] Force codec   pdhales at optusnet.com.au
• [Asterisk-Users] for review   Miles Scruggs
• [Asterisk-Users] Instant Message?   Steve Totaro
• [Asterisk-Users] wellgate registration 3802   Jolly M. Recto
• [Asterisk-Users] Instant Message?   Zhiqiang Li
• [Asterisk-Users] Uplink Skype2Sip   kevin ling
• [Asterisk-Users] Manager API Help   Darren Ellis
• [Asterisk-Users] Instant Message?   Kerry Garrison
• [Asterisk-Users] Planet VIP-320 DECT gateway with Asterisk?   kevin ling
• [Asterisk-Users] Force codec   Michael Strelnikov
• [Asterisk-Users] Instant Message?   wendell hamilton
• [Asterisk-Users] PRI Group Calling   Mark Edwards
• [Asterisk-Users] PRI Group Calling   Kevin P. Fleming
• [Asterisk-Users] Asterisk Dial Command Timeout not Accurate (not even close)   Peter J Dean
• [Asterisk-Users] wellgate registration 3802   Olle E Johansson
• [Asterisk-Users] Voipstunt, voipbuster, .... not working properly?   Ronald Wiplinger
• [Asterisk-Users] (no subject)   Tomislav Parčina
• [Asterisk-Users] question about DISA   Koopmann, Jan-Peter
• [Asterisk-Users] Voipstunt, voipbuster, .... not working properly?   Rudolf Ladyzhenskii
• [Asterisk-Users] Problems with registering iaxy   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Realtime oracle compiling problem   René Enskat [Teamware GmbH]
• [Asterisk-Users] Voipstunt, voipbuster, .... not working properly?   Ronald Wiplinger
• [Asterisk-Users] Re: update - 512 Simultaneous Calls with Digital Recording   Ben Dinnerville
• [Asterisk-Users] queue_log timestamp?   jan.sarin at securia.se
• [Asterisk-Users] CallerID   Jay Milk
• [Asterisk-Users] Manager API Help   Jon Farmer
• [Asterisk-Users] Voipstunt, voipbuster, .... not working properly?   Michiel van Baak
• [Asterisk-Users] Manager API Help   picciuX
• [Asterisk-Users] How to avoid "Avoiding initial deadlock...."   Joseph Rothstein
• [Asterisk-Users] Asterisk is not reconnecting   Michael Strelnikov
• [Asterisk-Users] G729a error   René Enskat [Teamware GmbH]
• [Asterisk-Users] Re: update - 512 Simultaneous Calls with Digital Recording   Luki
• [Asterisk-Users] quadBRI PCI ISDN on Suse Linux 10   Colin MacMillan
• [Asterisk-Users] Re: update - 512 Simultaneous Calls with DigitalRecording   Boris Bakchiev
• [Asterisk-Users] quadBRI PCI ISDN on Suse Linux 10   Dave Cotton
• [Asterisk-Users] Asterisk evaluating CLIP, then getting out of the way   Marc Rohlfing
• [Asterisk-Users] meetme   Giuseppe
• [Asterisk-Users] quadBRI PCI ISDN on Suse Linux 10   Colin MacMillan
• [Asterisk-Users] Sound Conversion Problems with Festival+Mbrola   Christian Gröger
• [Asterisk-Users] Call me for testing my system   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Call me for testing my system   Thomas Winter
• [Asterisk-Users] Call me for testing my system   hgaillac-sip at yahoo.fr
• [Asterisk-Users] WG: G729a error   René Enskat [Teamware GmbH]
• [Asterisk-Users] GXP-2000 phones stop registering   Gareth Blades
• [Asterisk-Users] Asterisk to CCM4 SIP Trunk one-way audio problem.   kevin ling
• [Asterisk-Users] AMP / Maintenance-Button missing   Thomas Broda
• [Asterisk-Users] GXP-2000 phones stop registering   Mark Edwards
• [Asterisk-Users] Asterisk Dial Command Timeout not Accurate (not even close)   Eric "ManxPower" Wieling
• [Asterisk-Users] Double Call Progress tones   kevin ling
• [Asterisk-Users] Best ATA for general residential deployment??   Rich Adamson
• [Asterisk-Users] Double Call Progress tones   Lee Archer
• [Asterisk-Users] Double Call Progress tones   Ron Wellsted
• [Asterisk-Users] New Skype<>SIP gateway   Whisker, Peter
• [Asterisk-Users] Double-ring tone   kevin ling
• [Asterisk-Users] GXP-2000 phones stop registering   Gareth Blades
• [Asterisk-Users] Call me for testing my system   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Cisco Callmanager integration with asterisk   kevin ling
• [Asterisk-Users] New Skype<>SIP gateway   Michael Graves
• [Asterisk-Users] Instant Message?   Kyle Sexton
• [Asterisk-Users] quadBRI PCI ISDN on Suse Linux 10   Colin MacMillan
• [Asterisk-Users] Asterisk stops responding when internet is down   Michael Strelnikov
• [Asterisk-Users] Call me for testing my system   Thomas Winter
• [Asterisk-Users] Re: need to make my oh323 work with quintum no gatekeeper   Goke Aruna
• [Asterisk-Users] Re: meetme   Tony Mountifield
• [Asterisk-Users] Asterisk stops responding when internet is down   Michiel van Baak
• [Asterisk-Users] sixtel   Curt Shaffer
• [Asterisk-Users] callerid name inboune from PRI   Steven
• [Asterisk-Users] sixtel   Steve Totaro
• [Asterisk-Users] AAstra 9133i register double account.. ??   William Harrison
• [Asterisk-Users] queue_log timestamp?   lenz
• [Asterisk-Users] Instant Message?   Stefan Reuter
• [Asterisk-Users] AMP / Maintenance-Button missing   Avi Miller
• [Asterisk-Users] RTP mixer in Asterisk   Leonardo (listas)
• [Asterisk-Users] Directory App() is running for a while, like blocked/freeze? in the same name...   Marco Mouta
• [Asterisk-Users] [asterisk-dev] RTP mixer in Asterisk   Leonardo (listas)
• [Asterisk-Users] G729a error   Kevin P. Fleming
• [Asterisk-Users] AMP / Maintenance-Button missing   Alex Robar
• [Asterisk-Users] call center running Asterisk - sound quality - critical!   Dov Bigio
• [Asterisk-Users] Re: Directory App() is running for a while, like blocked/freeze? in the same name...   Marco Mouta
• [Asterisk-Users] How to set AbsoluteTimeout for DirectoryApp() ? Is this the safest way?   Marco Mouta
• [Asterisk-Users] sixtel   Jay Milk
• [Asterisk-Users] Re: queue issue   Tomislav Parčina
• [Asterisk-Users] How to set AbsoluteTimeout for DirectoryApp() ? Is this the safest way?   Kevin P. Fleming
• [Asterisk-Users] Outbound calls through Broadvoice   Mike Raley
• [Asterisk-Users] sixtel   Zac Amsler
• [Asterisk-Users] asterisk box as a voip gateway   Mark Quitoriano
• [Asterisk-Users] RTP Timestamp errors   Erik
• [Asterisk-Users] RTP Timestamp errors   Kevin P. Fleming
• [Asterisk-Users] callerid name inboune from PRI   Alexander Lopez
• [Asterisk-Users] NORTH CAROLINA: Any interest in starting NC User Group?   Mike Clark
• [Asterisk-Users] SIP channel unavailable/busy/really not there   Steve Kennedy
• [Asterisk-Users] call center running Asterisk - sound quality - critical!   C F
• [Asterisk-Users] call center running Asterisk - sound quality - critical!   Colin MacMillan
• [Asterisk-Users] "chan_iax2.c: Ooh, voice format changed to ..."   Mimmus
• [Asterisk-Users] Asterisk/InterTel Axxess via MGCP? Anyone?   Shane DeRidder
• [Asterisk-Users] "chan_iax2.c: Ooh, voice format changed to ..."   Joshua Colp
• [Asterisk-Users] call center running Asterisk - sound quality -critical!   TC
• [Asterisk-Users] faxing setup   Corne Labuschagne
• [Asterisk-Users] faxing setup   Hermann Wecke
• [Asterisk-Users] Any Aheeva Users?   asterisk at dgtec.net.do
• [Asterisk-Users] App Page() in 1.2.5   Douglas Garstang
• [Asterisk-Users] faxing setup   Bartosz Jozwiak
• [Asterisk-Users] App Page() in 1.2.5   Kevin P. Fleming
• [Asterisk-Users] ANI and DNIS Seperation on a PRI (Telephony Numbering Plan (E.164/E.163) (1) '*4105556654*8005550215*' ])   Steve Totaro
• [Asterisk-Users] Outbound calls through Broadvoice   Ronald Wiplinger
• [Asterisk-Users] still no solution for me, if one provider fails.   Ronald Wiplinger
• [Asterisk-Users] Audio problems   Ronald Wiplinger
• [Asterisk-Users] still no solution for me, if one provider fails.   Derek Whitten
• [Asterisk-Users] App Page() in 1.2.5   Alexander Lopez
• [Asterisk-Users] ANI and DNIS Seperation on a PRI (Telephony Numbering Plan (E.164/E.163) (1) '*4105556654*8005550215*' ])   Alexander Lopez
• [Asterisk-Users] RE: still no solution for me, if one provider   Brent Torrenga
• [Asterisk-Users] Re: Outbound calls through Broadvoice   Mike Raley
• [Asterisk-Users] ANI and DNIS Seperation on a PRI (TelephonyNumbering Plan (E.164/E.163) (1) '*4105556654*8005550215*' ])   Steve Totaro
• [Asterisk-Users] ANI and DNIS Seperation on a PRI (TelephonyNumbering Plan (E.164/E.163) (1) '*4105556654*8005550215*' ])   Alexander Lopez
• [Asterisk-Users] RE: still no solution for me, if one provider   Ronald Wiplinger
• [Asterisk-Users] Re: callerid name inboune from PRI   Steven
• [Asterisk-Users] Texas User Group   Bruce Reeves
• [Asterisk-Users] Instant Message?   Bruce Reeves
• [Asterisk-Users] Asterisk to CCM4 SIP Trunk one-way audio problem.   Jeremy Wadhams
• [Asterisk-Users] E1 PRI problem with TE205P   Phone Dev
• [Asterisk-Users] Manager API Help   Jeremy Wadhams
• [Asterisk-Users] Re: callerid name inboune from PRI   Jerry Jones
• [Asterisk-Users] Problem - Voicemail resets phone   Paul A Brown
• [Asterisk-Users] RE: still no solution for me, if one   Brent Torrenga
• [Asterisk-Users] Steps to make trunked iax2   Andy Kuo
• [Asterisk-Users] Re: callerid name inboune from PRI   Alexander Lopez
• [Asterisk-Users] Problem - Voicemail resets phone   Alexander Lopez
• [Asterisk-Users] App Page() in 1.2.5   Douglas Garstang
• [Asterisk-Users] how to communicate two PCs on LAN with   Yusuf
• R: [Asterisk-Users] E1 PRI problem with TE205P   Phone Dev
• [Asterisk-Users] Telephony newbie need advice for integration Nortel MICS 4.1 with Asterisk via T1/E1 interface   CG
• [Asterisk-Users] E1 PRI problem with TE205P   Dennis Walker
• [Asterisk-Users] Steps to make trunked iax2   Rich Adamson
• [Asterisk-Users] Problem - Voicemail resets phone   Paul A Brown
• [Asterisk-Users] Problem - Voicemail resets phone   Peter Braidwood
• [Asterisk-Users] Queues - Dumb question   Marco Campos
• [Asterisk-Users] Asterisk and Cisco Callmanager   Terry Wade
• [Asterisk-Users] Polycom TOS   Jonathan k. Creasy
• [Asterisk-Users] App Page() in 1.2.5   John Novack
• [Asterisk-Users] Group funcations not functioning   Raymond Chen
• [Asterisk-Users] Re: Re: callerid name inboune from PRI   Steven
• R: [Asterisk-Users] E1 PRI problem with TE205P   Niklas Larsson
• [Asterisk-Users] Wanted any /all used out of service Digium boards Mark   VOICEIN at aol.com
• [Asterisk-Users] Wanted any /all used out of service Digium boards Mark   Joshua Colp
• [Asterisk-Users] Wanted any /all used out of service Digium boards Mark   Kevin P. Fleming
• [Asterisk-Users] Re: GXP-2000 phones stop registering   Lonnie Abelbeck
• [Asterisk-Users] Group funcations not functioning   Thomas Winter
• [Asterisk-Users] Problem with Asterisk and Grandstream HT286   Álvaro Palma
• [Asterisk-Users] Is this possible? (queue setup)   Andre Ruiz
• [Asterisk-Users] call center running Asterisk - sound quality -critical!   Rob Terhaar
• [Asterisk-Users] HTML / PHP   Jeremy
• [Asterisk-Users] HTML / PHP   Justin Tunney
• [Asterisk-Users] Re: Re: callerid name inboune from PRI   Steven
• [Asterisk-Users] pause / unpausequeuemember   Johann
• [Asterisk-Users] Asterisk Dial Command Timeout not Accurate (not even close)   Peter J Dean
• [Asterisk-Users] queue_log timestamp?   Johann
• [Asterisk-Users] te110p and interrupts   Anton Krall
• [Asterisk-Users] queue_log timestamp?   Anton Krall
• [Asterisk-Users] [ISSUE] Honouring Silent Caller ID Numbers   Peter J Dean
• [Asterisk-Users] te110p and interrupts   Infobox Peru
• [Asterisk-Users] Any Aheeva Users?   Infobox Peru
• [Asterisk-Users] ANI and DNIS Seperation on a PRI(TelephonyNumbering Plan (E.164/E.163) (1)'*4105556654*8005550215*' ])   Steve Totaro
• [Asterisk-Users] Routing SIP calls via URI   Jeremy Wadhams
• [Asterisk-Users] te110p and interrupts   Anton Krall
• [Asterisk-Users] te110p and interrupts   Mojo with Horan & Company, LLC
• [Asterisk-Users] te110p and interrupts   Boris Bakchiev
• [Asterisk-Users] te110p and interrupts   Anton Krall
• [Asterisk-Users] HTML / PHP   Jay Milk
• [Asterisk-Users] te110p and interrupts   Anton Krall
• [Asterisk-Users] Polycom TOS   Bill Michaelson
• [Asterisk-Users] te110p and interrupts   Mojo with Horan & Company, LLC
• [Asterisk-Users] Problems registering Linksys SPA941 with * via SIP   Damian Funnell
• [Asterisk-Users] te110p and interrupts   Eric "ManxPower" Wieling
• [Asterisk-Users] Any Aheeva Users?   Steve Totaro
• [Asterisk-Users] [ISSUE] Honouring Silent Caller ID Numbers   Steve Totaro
• [Asterisk-Users] Problems registering Linksys SPA941 with * via SIP   kevin ling
• [Asterisk-Users] App Page() in 1.2.5   kevin ling
• [Asterisk-Users] Asterisk and Cisco Callmanager   kevin ling
• 답장: [Asterisk-Users] Problems registering Linksys SPA941 with * via SIP   한정현
• [Asterisk-Users] App Page() in 1.2.5   John Novack (port)
• [Asterisk-Users] App Page() in 1.2.5   kevin ling
• [Asterisk-Users] callerid name inboune from PRI   Andres
• [Asterisk-Users] One digit too short dialed, stay for ever there in the dialplan!   Ronald Wiplinger
• [Asterisk-Users] Choppy Sound when using linux router or asterisk   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] One digit too short dialed, stay for ever there in the dialplan!   Kevin P. Fleming
• [Asterisk-Users] te110p and interrupts   Anton Krall
• [Asterisk-Users] Asterisk BRI in the USA   Mark Coccimiglio
• [Asterisk-Users] Beeps and noises during calls   Sean Garland
• [Asterisk-Users] Vertical   Mike Diehl (Encrypted email preferred)
• [Asterisk-Users] asterisk credit card processing   Joseph
• [Asterisk-Users] callerid name inboune from PRI   C F
• [Asterisk-Users] asterisk credit card processing   Jean-Michel Hiver
• [Asterisk-Users] Asterisk BRI in the USA   Lacy Moore - Aspendora
• [Asterisk-Users] One digit too short dialed, stay for ever there in the dialplan!   Ronald Wiplinger
• [Asterisk-Users] SPA-941/942 Bulk provisioning   Kerry Garrison
• [Asterisk-Users] Asterisk BRI in the USA   Krzysztof Drewicz
• [Asterisk-Users] asterisk credit card processing   pdhales at optusnet.com.au
• [Asterisk-Users] Bandwidth Management   Andy Tan
• [Asterisk-Users] asterisk credit card processing   Joseph
• [Asterisk-Users] Asterisk and Cisco Callmanager   Terry Wade
• [Asterisk-Users] RTP Timestamp errors   Erik
• [Asterisk-Users] Asterisk BRI in the USA   Mark Coccimiglio
• [Asterisk-Users] Database server   Dumpolid Exeplish
• [Asterisk-Users] Re: GXP-2000 phones stop registering   Gareth Blades
• [Asterisk-Users] SPA-941/942 Bulk provisioning   Gareth Blades
• [Asterisk-Users] Echo in some queues but not others   Joseph Rothstein
• [Asterisk-Users] Choppy Sound when using linux router or asterisk   Lacy Moore - Aspendora
• [Asterisk-Users] Vertical   Lacy Moore - Aspendora
• [Asterisk-Users] Re: meetme   Giuseppe
• [Asterisk-Users] [ISSUE] Honouring Silent Caller ID Numbers   Lacy Moore - Aspendora
• [Asterisk-Users] Native music on hold on 1.0   Tomislav Parčina
• [Asterisk-Users] Differences 1.0 vs. 1.2   Tomislav Parčina
• [Asterisk-Users] still no solution for me, if one provider fails.   Mimmus
• [Asterisk-Users] App Page() in 1.2.5   Alexander Lopez
• [Asterisk-Users] asterisk credit card processing   Alexander Lopez
• [Asterisk-Users] Major issue: More incompatible frame messages   Joseph Rothstein
• [Asterisk-Users] Database server   Barry Flanagan
• [Asterisk-Users] Queues - Dumb question   picciuX
• [Asterisk-Users] Native music on hold on 1.0   Gareth Blades
• [Asterisk-Users] Major issue: More incompatible frame messages   Joseph Rothstein
• [Asterisk-Users] the best billing tool for Asterisk   Joao Pereira
• [Asterisk-Users] Trial Version of Asterisk Interface Available   Vikram Rangnekar
• [Asterisk-Users] SPA-941/942 Bulk provisioning   Rich Adamson
• [Asterisk-Users] G726-40 required - Help!   Carsten Bock
• [Asterisk-Users] asterisk credit card processing   Sean Cook
• [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)   Jim Rice
• [Asterisk-Users] Trial Version of Asterisk Interface Available   Doug Lytle
• [Asterisk-Users] Asterisk BRI in the USA   Rich Adamson
• [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)   Avi Miller
• [Asterisk-Users] te110p and interrupts   Kenneth Lussier
• [Asterisk-Users] Automatic 3 Way Call   Shad Mortazavi
• [Asterisk-Users] callerid name inboune from PRI   Jerry Jones
• [Asterisk-Users] Screen macro - not working   Mike
• [Asterisk-Users] Choppy Sound when using linux router or asterisk   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Asterisk stops responding when internet is down   Michael Strelnikov
• [Asterisk-Users] Force codec   Michael Strelnikov
• [Asterisk-Users] te110p and interrupts   Andrew Kohlsmith
• [Asterisk-Users] One digit too short dialed, stay for ever there in the dialplan!   BJ Weschke
• [Asterisk-Users] Bandwidth Management   Andy Tan
• [Asterisk-Users] Asterisk is not reconnecting   Michael Strelnikov
• [Asterisk-Users] Question on clicking   Matt
• [Asterisk-Users] App Page() in 1.2.5   John Novack
• [Asterisk-Users] App Page() in 1.2.5   John Novack
• [Asterisk-Users] Vertical   Matt
• [Asterisk-Users] Re: Received VNAK: resending outstanding frames?   Carey O'Shea
• [Asterisk-Users] Why is the internet connection important to LAN and PSTN calls?   Brent Torrenga
• [Asterisk-Users] Night Mode and indicators   Jason Adams
• [Asterisk-Users] SPA-941/942 Bulk provisioning   Alberto Sagredo
• [Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??   Pimjai Wesnarat
• [Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??   Lee Archer
• AW: [Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??   Marcus.Rothe at bertelsmann.de
• [Asterisk-Users] SPA-941/942 Bulk provisioning   tracinet
• AW: [Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??   Pimjai Wesnarat
• [Asterisk-Users] Agent with multiple phones in multiple queues   Kristian Marcroft
• [Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??   Gareth Blades
• [Asterisk-Users] Question on clicking   Bob McDowell
• [Asterisk-Users] Logoff time of an agent.   Kristian Marcroft
• [Asterisk-Users] Night Mode and indicators   picciuX
• [Asterisk-Users] Virtual terminal running CLI   Aaron Daniel
• [Asterisk-Users] PRI outbound call error detection   Gareth Blades
• [Asterisk-Users] Vertical   Damon Estep
• [Asterisk-Users] Why is the internet connection important to LAN and PSTN calls?   picciuX
• [Asterisk-Users] Re: Virtual terminal running CLI   Aaron Daniel
• [Asterisk-Users] Question on clicking   Matt
• [Asterisk-Users] Question on clicking   Matt
• [Asterisk-Users] Re: Trial Version of Asterisk Interface Available   Vikram Rangnekar
• [Asterisk-Users] te110p and interrupts   Anton Krall
• [Asterisk-Users] te110p and interrupts   Anton Krall
• [Asterisk-Users] Bandwidth Management   Alexander Lopez
• [Asterisk-Users] G726-40 required - Help!   Joshua Colp
• [Asterisk-Users] the best billing tool for Asterisk   Joshua Colp
• [Asterisk-Users] Asterisk stops responding when internet is down   Joshua Colp
• [Asterisk-Users] Agent with multiple phones in multiple queues   Alexander Lopez
• [Asterisk-Users] XO Callerid NAME   Larry Linde
• [Asterisk-Users] Bandwidth Management   Rusty Dekema
• [Asterisk-Users] Snap for Asterisk   mitcheloc at gmail.com
• [Asterisk-Users] G726-40 required - Help!   Carsten Bock
• [Asterisk-Users] Asterisk BRI in the USA   Rusty Dekema
• [Asterisk-Users] ExternalIVR   Waldo Rubinstein
• [Asterisk-Users] callerid name inboune from PRI   C F
• [Asterisk-Users] log messages...   Dov Bigio
• [Asterisk-Users] Why is the internet connection important to LAN and PSTN calls?   Joseph Tanner
• [Asterisk-Users] Cisco 7970 SIP Config   Jeremiah Millay
• R: [Asterisk-Users] te110p and interrupts   Phone Dev
• [Asterisk-Users] nic aliases not working   Michael George
• R: R: [Asterisk-Users] E1 PRI problem with TE205P   Phone Dev
• [Asterisk-Users] te110p and interrupts   Begumisa Gerald M
• [Asterisk-Users] SPA-941/942 Bulk provisioning   mustardman29
• [Asterisk-Users] ExternalIVR   Kevin P. Fleming
• [Asterisk-Users] te110p and interrupts   Begumisa Gerald M
• [Asterisk-Users] Re: MACRO_RESULT=ABORT   Shaun
• [Asterisk-Users] Calls made through Manager API get same channel and "Unique" ID   Darren Ellis
• [Asterisk-Users] asterisk credit card processing   Mike Clark
• [Asterisk-Users] STUN Server info   Wasif
• [Asterisk-Users] HELP NEEDED: "odbc show" crashes Asterisk... and I have no idea of what is going on!!!   Leo Burd
• Fwd: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Matt Roth
• Fwd: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Matt Florell
• [Asterisk-Users] Cisco 7960 problems   Paul A Brown
• [Asterisk-Users] RE: Fatpipe Support - Authorization to open Box - fwrps2001101288   Tim Reimers
• Fwd: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Kevin P. Fleming
• [Asterisk-Users] TNT Max Config   mezzmor at aim.com
• [Asterisk-Users] Re: update - 512 Simultaneous Calls with Digital Recording   Matt Roth
• [Asterisk-Users] Re: update - 512 Simultaneous Calls with DigitalRecording   Matt Roth
• [Asterisk-Users] Why is the internet connection important to LAN and PSTN calls?   Andre Ruiz
• [Asterisk-Users] core dump...   Dov Bigio
• [Asterisk-Users] nic aliases not working   Daniel Hazelbaker
• [Asterisk-Users] still no solution for me, if one provider fails.   Peter J Dean
• [Asterisk-Users] chan_btp: dialing external phone number when bluetooth not present?   Mike Garey
• [Asterisk-Users] call center running Asterisk - sound quality - critical!   Matt Roth
• [Asterisk-Users] res_config_mysql.so: undefined symbol: __stack_chk_fail   kritikus Araklidas
• [Asterisk-Users] asterisk credit card processing   Joseph
• [Asterisk-Users] callerid name inboune from PRI   Greg Oliver
• [Asterisk-Users] Re: Received VNAK: resending outstanding frames?   Tim Panton
• [Asterisk-Users] Re: Received VNAK: resending outstanding frames?   Tim Panton
• [Asterisk-Users] still no solution for me, if one provider fails.   Noah Miller
• [Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??   Tim Panton
• [Asterisk-Users] Re: Received VNAK: resending outstanding frames?   Carey O'Shea
• [Asterisk-Users] Re: Hangupcause is not enough on PRI   Pibix
• [Asterisk-Users] E1 Disconnection Asterisk behind an old PBX   Marco Mouta
• [Asterisk-Users] Texas User Group   Ryan Burke
• [Asterisk-Users] Re: update - 512 Simultaneous Callswith DigitalRecording   Boris Bakchiev
• [Asterisk-Users] Texas User Group   Lacy Moore - Aspendora
• [Asterisk-Users] Cisco 7960 6.3 unlock/reset?   Shaun
• [Asterisk-Users] Texas User Group   Bruce Reeves
• [Asterisk-Users] nic aliases not working   Michael George
• [Asterisk-Users] Texas User Group   Tom Chandler
• [Asterisk-Users] Cisco 7960 6.3 unlock/reset?   Tim Connolly
• [Asterisk-Users] call center running Asterisk - sound quality- critical!   Wai Wu
• [Asterisk-Users] Texas User Group   Aaron Daniel
• [Asterisk-Users] Performance: Xeon or Opteron?   Tim Connolly
• [Asterisk-Users] res_config_mysql.so: undefined symbol: __stack_chk_fail   Tim Connolly
• [Asterisk-Users] call center running Asterisk - sound quality- critical!   C F
• [Asterisk-Users] AstriCon Update: Europe Early Bird Ends Saturday   Steven Sokol
• [Asterisk-Users] Bandwidth Management   Andy Tan
• [Asterisk-Users] res_config_mysql.so: undefinedsymbol: __stack_chk_fail   kritikus Araklidas
• [Asterisk-Users] Performance: Xeon or Opteron?   Waldo Rubinstein
• [Asterisk-Users] How to config firewall for RTP/RTCP?   谢水全
• [Asterisk-Users] Re: update - 512 Simultaneous Callswith DigitalRecording   Leo Ann Boon
• [Asterisk-Users] Bandwidth Management   Jean-Michel Hiver
• [Asterisk-Users] TE410P upgrade to TE411P => (solution to) no more fax carrier detection !   tony.robin at free.fr
• [Asterisk-Users] Polycom SIP 1.6.5 reloading   James Andrewartha
• [Asterisk-Users] SPA-3000 call pickup behind a PABX   Dieter Jansen
• [Asterisk-Users] Snap for Asterisk   Bartosz Piec
• [Asterisk-Users] Asterisk stops responding when internet is down   Olle E Johansson
• [Asterisk-Users] Why is the internet connection important to LAN and PSTN calls?   Olle E Johansson
• [Asterisk-Users] Asterisk stops responding when internet is down   Michael Strelnikov
• [Asterisk-Users] Trial Version of Asterisk Interface Available   Olle E Johansson
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   MBIT Technologies
• [Asterisk-Users] Asterisk stops responding when internet is down   Olle E Johansson
• [Asterisk-Users] Re: Received VNAK: resending outstanding frames?   Tim Panton
• [Asterisk-Users] still no solution for me, if one provider fails.   Mimmus
• [Asterisk-Users] Why is the internet connection important to LAN and PSTN calls?   Cristian Draghici
• [Asterisk-Users] Asterisk stops responding when internet is down   Michael Strelnikov
• [Asterisk-Users] Cisco 7960 6.3 unlock/reset?   Joseph Rothstein
• [Asterisk-Users] Re: Why is the internet connection important to LAN and PSTN calls?   Benny Amorsen
• [Asterisk-Users] SipXPhone   Mark Hayward
• [Asterisk-Users] Where is the difference sip.conf - Real-time ?   Ronald Wiplinger
• [Asterisk-Users] call center running Asterisk - sound quality- critical!   Tamas
• [Asterisk-Users] SIP channel unavailable/busy/really not there   Peter Fern
• [Asterisk-Users] iax2 show netstats   yusuf
• [Asterisk-Users] Where is the difference sip.conf - Real-time ?   Alban
• [Asterisk-Users] billing with PostgreSQL   Joao Pereira
• [Asterisk-Users] Re: Received VNAK: resending outstanding frames?   Carey O'Shea
• [Asterisk-Users] SIP call hangup from asterisk CLI   Abhimanyu Rapria
• [Asterisk-Users] Where is the difference sip.conf - Real-time ?   Ronald Wiplinger
• [Asterisk-Users] Failed to recieve Fax: Asterisk - IAXModem - Hylafax   Pimjai Wesnarat
• [Asterisk-Users] help -- voicemail   chan (Alpha Trilogies Networks)
• [Asterisk-Users] Asterisk BRI in the USA   Michael Graves
• [Asterisk-Users] SIP call hangup from asterisk CLI   Marco Mouta
• [Asterisk-Users] free video (soft) phone available?   Ronald Wiplinger
• [Asterisk-Users] Texas User Group   Michael Graves
• [Asterisk-Users] g.726 codec not working in one direction   Thomas Winter
• [Asterisk-Users] Snap for Asterisk   mitcheloc at gmail.com
• [Asterisk-Users] Asterisk BRI in the USA   Steve Brown
• [Asterisk-Users] Oh323 inband DTMF   Tomislav Parčina
• [Asterisk-Users] Why is the internet connection important to LAN and PSTN calls?   Olle E Johansson
• [Asterisk-Users] TE410P upgrade to TE411P => (solution to) no more fax carrier detection !   Kevin P. Fleming
• [Asterisk-Users] queue_log timestamp?   Tomas Stribrny
• [Asterisk-Users] SIP conections, with RTP not going trough Asterisk   Tiago Stein D`Agostini
• [Asterisk-Users] SIP conections, with RTP not going trough Asterisk   Ronald Wiplinger
• [Asterisk-Users] Asterisk BRI in the USA   Joe Greco
• [Asterisk-Users] DID'S Romania - Bucharest   Oliver Vermeulen
• [Asterisk-Users] Texas User Group   Greg Camp
• [Asterisk-Users] Automatic 3 Way Call   Shad Mortazavi
• [Asterisk-Users] DID'S Romania - Bucharest   Roger Schreiter
• [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone   Steve Feinstein
• [Asterisk-Users] billing with PostgreSQL   Andy Tan
• [Asterisk-Users] Performance: Xeon or Opteron?   RandyW
• [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone   BJ Weschke
• [Asterisk-Users] help -- voicemail   Melcon Moraes
• [Asterisk-Users] Performance: Xeon or Opteron?   John Novack
• [Asterisk-Users] Trunking Protocols   Andy Tan
• [Asterisk-Users] Macro-hangupcall - has a Wait(5) - Ast@Home --- why?   Marco Mouta
• [Asterisk-Users] GXP-2000 phones stop registering   Gareth Blades
• [Asterisk-Users] Texas User Group   Bruce Reeves
• [Asterisk-Users] Performance: Xeon or Opteron?   Rich Adamson
• [Asterisk-Users] Bandwidth Management   Wai Wu
• [Asterisk-Users] Performance: Xeon or Opteron?   Kristian Kielhofner
• [Asterisk-Users] SipXPhone   Greg Camp
• [Asterisk-Users] call center running Asterisk - sound quality-critical!   Wai Wu
• [Asterisk-Users] Newbie MOH and call transfer question   kevin ling
• [Asterisk-Users] Performance: Xeon or Opteron?   Rich Adamson
• [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Henri Herscher
• [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone   Steve Feinstein
• [Asterisk-Users] call center running Asterisk - sound quality - critical!   Henri Herscher
• [Asterisk-Users] Performance: Xeon or Opteron?   John Novack
• [Asterisk-Users] Texas User Group   Aaron Daniel
• [Asterisk-Users] call center running Asterisk - sound quality-critical!   Matt Roth
• [Asterisk-Users] CAPI Installation Eicon Diva Server   nkohl at maninvestments.com
• [Asterisk-Users] Setting Codecs on the Fly   Douglas Garstang
• [Asterisk-Users] call center running Asterisk - sound quality-critical!   Kevin P. Fleming
• [Asterisk-Users] iax2 show netstats   Benchev
• [Asterisk-Users] TE410P upgrade to TE411P => (solution to) no more fax carrier detection !   Rob Lith
• [Asterisk-Users] TE410P upgrade to TE411P => (solution to) no more fax carrier detection !   Kevin P. Fleming
• [Asterisk-Users] call center running Asterisk - sound quality-critical!   Tamas
• [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Waldo Rubinstein
• [Asterisk-Users] call center running Asterisk - sound quality-critical!   Kevin P. Fleming
• [Asterisk-Users] * 1.2.4 & 1.2.6: "Ringing" anamoly   Ronald Lewis
• [Asterisk-Users] Setting Codecs on the Fly   Julio Arruda
• [Asterisk-Users] Setting Codecs on the Fly   Douglas Garstang
• [Asterisk-Users] Setting Codecs on the Fly   Alexander Lopez
• [Asterisk-Users] Bandwidth Management   Alexander Lopez
• [Asterisk-Users] Polycom VLANs   Rob Terhaar
• [Asterisk-Users] playback soundfile stored in mysql database   Akpome Akpoguma
• [Asterisk-Users] call center running Asterisk -sound quality-critical!   Wai Wu
• [Asterisk-Users] playback soundfile stored in mysql database   Alexander Lopez
• [Asterisk-Users] Polycom VLANs   BJ Weschke
• [Asterisk-Users] call center running Asterisk -sound quality-critical!   BJ Weschke
• [Asterisk-Users] Recording queue transfers   Maximiliano J. Goldsmid
• [Asterisk-Users] call center running Asterisk -sound quality-critical!   Tamas
• [Asterisk-Users] Macro-hangupcall - has a Wait(5) - Ast@Home --- why?   BJ Weschke
• [Asterisk-Users] Config with TE210P, Asterisk and Legacy PBX and FreePBX?   Remco Barende
• [Asterisk-Users] call center running Asterisk-sound quality-critical!   Wai Wu
• [Asterisk-Users] call center running Asterisk -sound quality-critical!   Matt Roth
• [Asterisk-Users] * 1.2.4 & 1.2.6: "Ringing" anamoly   Chris Shaw
• [Asterisk-Users] Callback Agents and Dial 'g' option   Johann
• [Asterisk-Users] Cisco 7960 won't dial (sccp)   shawnl at up.net
• [Asterisk-Users] playback soundfile stored in mysql database   Akpome Akpoguma
• [Asterisk-Users] free video (soft) phone available?   Mojo with Horan & Company, LLC
• [Asterisk-Users] DUNDi with SIP   Adam Robins
• [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone   Kyle Sexton
• [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone   Steve Feinstein
• [Asterisk-Users] Asterisk BRI in the USA   Mark Coccimiglio
• [Asterisk-Users] Polycom VLANs   Jerry Jones
• [Asterisk-Users] SIP MWI   David Gomillion
• [Asterisk-Users] Cisco 7960 won't dial (sccp)   Sergio Chersovani
• [Asterisk-Users] Call Forward and AGI   Jon Farmer
• [Asterisk-Users] SIP conections, with RTP not going trough Asterisk   Olle E Johansson
• [Asterisk-Users] Recording queue transfers   lenz
• [Asterisk-Users] SIP MWI   Olle E Johansson
• [Asterisk-Users] Company List   Bruce Reeves
• [Asterisk-Users] Playback sound file while on-line   Andre Courchesne - Consultant
• [Asterisk-Users] Company List   Curt Shaffer
• [Asterisk-Users] Texas User Group   JR Richardson
• [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)   Jim Rice
• [Asterisk-Users] Company List   Rich Adamson
• [Asterisk-Users] Company List   Matt Roth
• [Asterisk-Users] Cisco 7960 won't dial (sccp)   shawnl at up.net
• [Asterisk-Users] Company List   Curt Shaffer
• [Asterisk-Users] Company List   Bruce Reeves
• [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)   Andrew Kohlsmith
• [Asterisk-Users] CAPI Installation Eicon Diva Server   Avi Miller
• [Asterisk-Users] Asterisk BRI in the USA   Walt Reed
• [Asterisk-Users] Company List   Aaron Daniel
• [Asterisk-Users] Asterisk BRI in the USA   Alexander Lopez
• [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)   Jim Rice
• [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)   Andrew Kohlsmith
• [Asterisk-Users] Company List   C F
• [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)   Avi Miller
• [Asterisk-Users] Cisco 7960 won't dial (sccp)   Lacy Moore - Aspendora
• [Asterisk-Users] Texas User Group   Bruce Reeves
• [Asterisk-Users] Texas User Group   Linda Robertson
• [Asterisk-Users] Company List   Bruce Reeves
• [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)   Jim Rice
• [Asterisk-Users] Call Forward and AGI   Josh McAllister
• [Asterisk-Users] Asterisk 1.2.7 Released!   Asterisk Development Team
• [Asterisk-Users] Asterisk BRI in the USA   Leo Ann Boon
• [Asterisk-Users] ASterisk Back2back   Dirgan Putra
• [Asterisk-Users] Text labels on incoming call appearances?   William P.N. Smith
• [Asterisk-Users] help -- voicemail   El Flynn
• [Asterisk-Users] Multiple phones in same call   Rudolf Ladyzhenskii
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 21, Issue 70   chan (Alpha Trilogies Networks)
• [Asterisk-Users] SPA-941/942 Bulk provisioning   asterisk at anime.net
• [Asterisk-Users] SPA-941/942 Bulk provisioning   Ronald Wiplinger
• [Asterisk-Users] need help   Dirgan Putra
• [Asterisk-Users] Re: update - 512 Simultaneous Callswith DigitalRecording   asterisk at anime.net
• [Asterisk-Users] How to terminate ringing call before it is answered   Obelix
• [Asterisk-Users] freepbx dialing prefix   Sean Garland
• [Asterisk-Users] How to terminate ringing call before it is answered   Darren Wiebe
• [Asterisk-Users] freepbx dialing prefix   Kerry Garrison
• [Asterisk-Users] Problem with Voice Quality   mkumar at mantragroup.com
• [Asterisk-Users] Announcement: New Texas User Group formed   Bruce Reeves
• Fwd: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Ira
• [Asterisk-Users] Cisco 7960 won't dial (sccp)   Sergio Chersovani
• [Asterisk-Users] Re: Double sip logins   nik600
• [Asterisk-Users] AAstra 9133i register double account.. ??   nik600
• [Asterisk-Users] playback soundfile in memory   Akpome Akpoguma
• [Asterisk-Users] * 1.2.4 & 1.2.6: "Ringing" anamoly   Olle E Johansson
• [Asterisk-Users] Problem with Voice Quality   Leo Ann Boon
• [Asterisk-Users] Urgent: Unable To Execute after updating from SVN   Min Hwan Chang
• [Asterisk-Users] TE410P upgrade to TE411P => (solution to) no more fax carrier detection !   George Pajari
• [Asterisk-Users] AAstra 9133i register double account.. ??   Dave Cotton
• [Asterisk-Users] Re: update - 512 Simultaneous Callswith DigitalRecording   Leo Ann Boon
• [Asterisk-Users] Asterisk BRI in the USA   David Waugh
• [Asterisk-Users] web meetme instructions   Ben Q
• [Asterisk-Users] Background music in call   Hüseyin
• [Asterisk-Users] Asterisk BRI in the USA   Mark Coccimiglio
• [Asterisk-Users] voicemail use external smtp server for sending mail   nik600
• [Asterisk-Users] Any way to prevent this from happening   Joseph Rothstein
• [Asterisk-Users] free tollfree termination   Gustavo Hernandez
• [Asterisk-Users] Music on hold problem   Daniel Korndorfer
• [Asterisk-Users] Music on hold problem   Gareth Blades
• [Asterisk-Users] Hangupcause to handle Called party disconnect ? PSTN----E1----OldPBX---E1--Asterisk   Marco Mouta
• [Asterisk-Users] ast_sched_runq ran 281 scheduled tasks all at once   Gareth Blades
• [Asterisk-Users] Music on hold problem   Daniel Korndorfer
• [Asterisk-Users] IP logging   Andrew Nowrot
• [Asterisk-Users] How to terminate ringing call before it is answered?   Obelix
• [Asterisk-Users] Asterisk BRI in the USA   Walt Reed
• [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)   Andrew Kohlsmith
• [Asterisk-Users] OT: MWI on Treo 600/650   David Cook
• [Asterisk-Users] Asterisk BRI in the USA   Joe Greco
• [Asterisk-Users] OT: MWI on Treo 600/650   Andrew Kohlsmith
• [Asterisk-Users] AAstra 9133i register double account.. ??   William Harrison
• [Asterisk-Users] IP logging   William Piper
• [Asterisk-Users] AgentCalled event   Alex Brett
• [Asterisk-Users] Music on hold problem   Don Pobanz
• [Asterisk-Users] Question on parkinglot   Matt
• [Asterisk-Users] Codec GSM Makefile Patch for IA64   Steve Totaro
• [Asterisk-Users] AgentCalled event   BJ Weschke
• [Asterisk-Users] IP logging   Andrew Nowrot
• [Asterisk-Users] freepbx dialing prefix   Sean Garland
• [Asterisk-Users] sip nat bug   marek cervenka
• [Asterisk-Users] Music on hold problem   Alex Brett
• [Asterisk-Users] (no subject)   Steve Totaro
• [Asterisk-Users] app_meetme.so   Sebastian Milioto
• [Asterisk-Users] free tollfree termination   Waldo Rubinstein
• [Asterisk-Users] fxotune error   Giorgio Incantalupo
• [Asterisk-Users] app_meetme.so   Steve Totaro
• [Asterisk-Users] Music on hold problem   Justin Tunney
• [Asterisk-Users] app_meetme.so   Sebastian Milioto
• [Asterisk-Users] app_meetme.so   Steve Totaro
• [Asterisk-Users] sip nat bug   Kevin P. Fleming
• [Asterisk-Users] OT: MWI on Treo 600/650   Mike Dent
• [Asterisk-Users] TE410P upgrade to TE411P => (solution to) no more fax carrier detection !   Kevin P. Fleming
• [Asterisk-Users] NAT/STUN Server   Wasif
• [Asterisk-Users] Re: OT: MWI on Treo 600/650   David Cook
• [Asterisk-Users] Display "Confideltial" or "unknown" on called id display   Andre Courchesne - Consultant
• [Asterisk-Users] playback soundfile in memory   Matt Roth
• [Asterisk-Users] Music on hold problem   Don Pobanz
• [Asterisk-Users] Display "Confideltial" or "unknown" on called id display   Rich Adamson
• [Asterisk-Users] callerid name inboune from PRI   Matthew Fredrickson
• [Asterisk-Users] Display "Confideltial" or "unknown" on called iddisplay   kevin ling
• [Asterisk-Users] Display "Confideltial" or "unknown" on called iddisplay   Steve Totaro
• [Asterisk-Users] Display "Confideltial" or "unknown" on called id display   Kevin P. Fleming
• [Asterisk-Users] Display "Confideltial" or "unknown" on called id display   Andrew Latham
• [Asterisk-Users] Display "Confideltial" or "unknown" on calledid display   Steve Totaro
• [Asterisk-Users] NAT/STUN Server   William Piper
• [Asterisk-Users] Segfault on Inbound call?   Matt
• [Asterisk-Users] Music on hold problem   Matt Roth
• [Asterisk-Users] Display "Confideltial" or "unknown" on callediddisplay   William Piper
• [Asterisk-Users] DTMF Not working for only one number   Aaron Daniel
• [Asterisk-Users] ast_sched_runq ran 281 scheduled tasks all at once   Andy Kuo
• [Asterisk-Users] Display "Confideltial" or "unknown" on called id display   Rich Adamson
• [Asterisk-Users] placing call with agi   Jon-o Addleman
• [Asterisk-Users] Sipura 2100   Darrell Long
• [Asterisk-Users] SIP register question   Steven Ringwald
• [Asterisk-Users] Asterisk 1.2.7 Page()   Douglas Garstang
• [Asterisk-Users] call center running Asterisk -sound quality-critical!   Matt Roth
• [Asterisk-Users] Set language in Asterisk auto-dial out   Benoît Mérouze
• [Asterisk-Users] Help Cas Circuit   Ali Arshad
• [Asterisk-Users] Asterisk 1.2.7 Page()   Kevin P. Fleming
• [Asterisk-Users] Ztmonitor shows RX is always on.   Min Hwan Chang
• [Asterisk-Users] AAstra 9133i register double account.. ??   Matt
• [Asterisk-Users] Asterisk 1.2.7.1 Released   Asterisk Development Team
• [Asterisk-Users] Early Media Enable?   Mohammed Salim
• [Asterisk-Users] Static on ZAP channels   Tim Jackson
• [Asterisk-Users] Static on ZAP channels   Kevin P. Fleming
• [Asterisk-Users] Static on ZAP channels   BJ Weschke
• [Asterisk-Users] call center running Asterisk -soundquality-critical!   Wai Wu
• [Asterisk-Users] Early Media Enable?   Nabeel Jafferali
• [Asterisk-Users] CANADA 911 Update   Bob's Leaky News Service
• [Asterisk-Users] transforming g729 files to wav files   Kristian Kielhofner
• [Asterisk-Users] Asterisk 1.2.7 Page()   Josué Conti
• [Asterisk-Users] Asterisk 1.2.7 Page()   Douglas Garstang
• [Asterisk-Users] DTMF sensitivity   William Piper
• [Asterisk-Users] call center running Asterisk-soundquality-critical!   Wai Wu
• AW: [Asterisk-Users] NAT/STUN Server   Till Stoermer
• [Asterisk-Users] Music on hold problem   Don Pobanz
• [Asterisk-Users] spa-942 support Page() / Intercom correctly?   Rich Adamson
• [Asterisk-Users] Performance: Xeon or Opteron?   Anton Krall
• [Asterisk-Users] SIP/ShoreTel REFER support   Magnus Kelly
• [Asterisk-Users] Asterisk codec detection problem !   Hamid Hashemi
• [Asterisk-Users] connecting Digium E1 pri card to panasonic TD-500   Krzysztof Drewicz
• [Asterisk-Users] Sipura 2100   Mike Diehl (Encrypted email preferred)
• [Asterisk-Users] Performance: Xeon or Opteron?   Rich Adamson
• [Asterisk-Users] Performance: Xeon or Opteron?   John Novack
• [Asterisk-Users] NAT/STUN Server   Ron Senykoff
• [Asterisk-Users] Anyone played with app_amd?   Josh McAllister
• [Asterisk-Users] automated response   abornand
• [Asterisk-Users] Anyone played with app_amd?   Kevin P. Fleming
• [Asterisk-Users] Announcement: New Texas User Group formed   Ryan Burke
• [Asterisk-Users] Set language in Asterisk auto-dial out   Benoît Mérouze
• [Asterisk-Users] What is Multi-layerAccess control   Carey Mould
• [Asterisk-Users] web meetme instructions   Dan Austin
• [Asterisk-Users] [asterisk-dev] RTP mixer in Asterisk   Mark Phillips
• [Asterisk-Users] Announcement: New Texas User Group formed   Bruce Reeves
• [Asterisk-Users] Digium cards, so disappointing !   Tony ROBIN
• [Asterisk-Users] Digium cards, so disappointing !   isamar at isamarmaia.org
• [Asterisk-Users] Digium cards, so disappointing !   Aaron Daniel
• [Asterisk-Users] Segfault on Inbound call?   John Novack
• [Asterisk-Users] Digium cards, so disappointing !   Lee Howard
• [Asterisk-Users] Anyone played with app_amd?   Matt Florell
• [Asterisk-Users] Will VoIP ITSP's be Next?   Bob's Leaky News Service
• [Asterisk-Users] Will VoIP ITSP's be Next?   Anthony Rodgers
• [Asterisk-Users] Will VoIP ITSP's be Next?   Wai Wu
• [Asterisk-Users] Will VoIP ITSP's be Next?   John Novack
• [Asterisk-Users] How to terminate ringing call before it is answered?   JP Carballo
• [Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. ???? Xlite   Ronald Wiplinger
• [Asterisk-Users] troubles with a Gateway audiocode (Mp104 fxs)   Ing. Rodrigo Fernandez Rodriguez
• [Asterisk-Users] What is Multi-layerAccess control   Rusty Dekema
• [Asterisk-Users] Asterisk no sound from sound card   Gary Hodder
• [Asterisk-Users] Asterisk no sound from sound card   Kerry Garrison
• [Asterisk-Users] HELP! Bad sound quality   Rudolf Ladyzhenskii
• [Asterisk-Users] Will VoIP ITSP's be Next?   Steve Totaro
• [Asterisk-Users] HELP! Bad sound quality   Tzafrir Cohen
• [Asterisk-Users] HELP! Bad sound quality   Rudolf Ladyzhenskii
• [Asterisk-Users] Cisco 7970 SIP   Tomislav Parčina
• [Asterisk-Users] How to cross compile asterisk for Axis ETRAX 100LX foxboard embedded device on Debian   Francois-Xavier Bas
• [Asterisk-Users] Asterisk hardware for new office suggestion   Simone
• [Asterisk-Users] Segfault on Inbound call?   Matt
• [Asterisk-Users] Digium cards, so disappointing !   Matt
• [Asterisk-Users] Segfault on Inbound call?   Rich Adamson
• [Asterisk-Users] Ztmonitor shows RX is always on.   Kyle Sexton
• [Asterisk-Users] Performance: Xeon or Opteron?   Anton Krall
• [Asterisk-Users] Digium cards, so disappointing !   Anton Krall
• [Asterisk-Users] Digium cards, so disappointing !   Anton Krall
• [Asterisk-Users] Digium cards, so disappointing !   Anton Krall
• [Asterisk-Users] Digium cards, so disappointing !   Matt Riddell (IT)
• [Asterisk-Users] tdm2400p and asterisk 1.2.1   Giorgio Incantalupo
• [Asterisk-Users] Asterisk hardware for new office suggestion   Rich Adamson
• [Asterisk-Users] tdm2400p and asterisk 1.2.1   Kevin P. Fleming
• [Asterisk-Users] Performance: Xeon or Opteron?   Rich Adamson
• [Asterisk-Users] Asterisk hardware for new office suggestion   Tim Panton
• [Asterisk-Users] Digium cards, so disappointing !   Rich Adamson
• [Asterisk-Users] Segfault on Inbound call?   Matt
• [Asterisk-Users] Will VoIP ITSP's be Next?   Alexander Lopez
• [Asterisk-Users] Digium cards, so disappointing !   Rich Adamson
• [Asterisk-Users] tdm2400p and asterisk 1.2.1   Giorgio Incantalupo
• [Asterisk-Users] asterisk with bluetooth headset howto   Andreas Nitsche
• [Asterisk-Users] change/toggle flash operator panel components   Giorgio Incantalupo
• [Asterisk-Users] asterisk with bluetooth headset howto   Alexander Lopez
• [Asterisk-Users] Digium cards, so disappointing !   Rusty Dekema
• [Asterisk-Users] tdm2400p and asterisk 1.2.1   Kevin P. Fleming
• [Asterisk-Users] Digium cards, so disappointing !   Kevin P. Fleming
• [Asterisk-Users] asterisk with bluetooth headset howto   Andreas Nitsche
• [Asterisk-Users] Digium cards, so disappointing !   Rich Adamson
• [Asterisk-Users] still no solution for me, if one provider fails.   Ronald Wiplinger
• [Asterisk-Users] Work available - India   Sahil Gupta
• [Asterisk-Users] A weekend of upgrade is coming for me - any hints?   Ronald Wiplinger
• [Asterisk-Users] Callback Agents and Dial 'g' option   Johann
• [Asterisk-Users] Packet Testing   Kevin Smith
• [Asterisk-Users] Background music in call   nzrh
• [Asterisk-Users] A weekend of upgrade is coming for me - any hints?   Rich Adamson
• [Asterisk-Users] Unicall and Fax   Carlos Chavez
• [Asterisk-Users] Digium cards, so disappointing !   Ryan Amos
• [Asterisk-Users] Polycom 501 resource full problems ...   phil.dawson at marnock.com
• [Asterisk-Users] Polycom 501 resource full problems ...   stoffell
• [Asterisk-Users] Polycom 501 resource full problems ...   Michael Welter
• [Asterisk-Users] the best billing tool for Asterisk   Guillermo Salas M.
• [Asterisk-Users] tdm2400p and asterisk 1.2.1   Giorgio Incantalupo
• [Asterisk-Users] Segfault on Inbound call?   Dinesh Nair
• [Asterisk-Users] tdm2400p and asterisk 1.2.1   Kevin P. Fleming
• [Asterisk-Users] Asterisk stops responding when internet is down   Jay Milk
• [Asterisk-Users] Asterisk stops responding when internet is down   Joshua Colp
• [Asterisk-Users] Digium cards, so disappointing !   Joshua Colp
• [Asterisk-Users] Packet Testing   Joshua Colp
• [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax   Thomas Artner
• [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax   Rob Terhaar
• [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax   Thomas Artner
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 21, Issue 81   Martin
• [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax   Thomas Artner
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 21, Issue 81   Aaron Daniel
• [Asterisk-Users] call center running Asterisk -soundquality-critical!   Matt Roth
• [Asterisk-Users] How to get 1.2.7 asterisk   Wai Wu
• [Asterisk-Users] call center running Asterisk-soundquality-critical!   Wai Wu
• [Asterisk-Users] How to get 1.2.7 asterisk   Matteo Brancaleoni
• [Asterisk-Users] attended transfer issue   Thomas Artner
• [Asterisk-Users] How to get 1.2.7 asterisk   Joshua Colp
• [Asterisk-Users] attended transfer issue   Michael Collins
• [Asterisk-Users] attended transfer issue   John Novack
• [Asterisk-Users] attended transfer issue   Jerry Jones
• [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax   Melcon Moraes
• [Asterisk-Users] attended transfer issue   John Novack
• [Asterisk-Users] Digium cards, so disappointing !   Jeff Gustafson
• [Asterisk-Users] attended transfer issue   Jerry Jones
• [Asterisk-Users] attended transfer issue   Michael Collins
• [Asterisk-Users] attended transfer issue   Rob Terhaar
• [Asterisk-Users] asterisk with bluetooth headset howto   Rob Terhaar
• [Asterisk-Users] Re: Cisco 7960 6.3 unlock/reset?   R.Trommer
• [Asterisk-Users] billing with PostgreSQL   Mindaugas Kezys
• [Asterisk-Users] attended transfer issue   Damon Estep
• [Asterisk-Users] asterisk or ser   Xaji Gaid
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Jeff Gustafson
• [Asterisk-Users] the best billing tool for Asterisk   Mindaugas Kezys
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Kevin P. Fleming
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Aaron Daniel
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Jeff Gustafson
• [Asterisk-Users] attended transfer issue   John Novack
• [Asterisk-Users] attended transfer issue   Thomas Artner
• [Asterisk-Users] asterisk 1.2.7.1 and app_rxfax   Thomas Artner
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Jeff Gustafson
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, sodisappointing !)   Jesus M. Lares
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Kevin P. Fleming
• [Asterisk-Users] attended transfer issue   Damon Estep
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Kevin P. Fleming
• [Asterisk-Users] Bluetooth (chan_btp): dialing external phone number through BTP/Zap when bluetooth device not present?   Mike Garey
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Jeff Gustafson
• [Asterisk-Users] My consulting story   Voce Lavoce
• [Asterisk-Users] My consulting story   Douglas Garstang
• [Asterisk-Users] My consulting story   Voce Lavoce
• [Asterisk-Users] My consulting story   Technical Support
• [Asterisk-Users] 7941/61 IP Phone SIP phone load - for CCM v5.0   Josh Reineke
• [Asterisk-Users] Ztmonitor shows RX is always on. FIXED.   Min Hwan Chang
• [Asterisk-Users] My consulting story   Michael Welter
• [Asterisk-Users] Digium cards, so disappointing !   Anton Krall
• [Asterisk-Users] Performance: Xeon or Opteron?   Anton Krall
• [Asterisk-Users] Digium cards, so disappointing !   Anton Krall
• [Asterisk-Users] still no solution for me, if one provider fails.   Peter J Dean
• [Asterisk-Users] Digium cards, so disappointing !   Anton Krall
• [Asterisk-Users] Unicall and Fax   Anton Krall
• [Asterisk-Users] My consulting story   Jay Milk
• [Asterisk-Users] My consulting story   Steve Totaro
• [Asterisk-Users] My consulting story   Steve Totaro
• [Asterisk-Users] Outgoing Ringback Indications IAX vs. SIP   Steve Feinstein
• [Asterisk-Users] Performance: Xeon or Opteron?   Matt
• [Asterisk-Users] Unicall and Fax   Paulo Scardine
• [Asterisk-Users] Digium cards, so disappointing !   George Pajari
• [Asterisk-Users] Digium cards, so disappointing !   Tim Panton
• [Asterisk-Users] Cisco 7960 International   Shaun
• [Asterisk-Users] rpms updated to 1.2.7.1 (was: Asterisk 1.2.7.1 Released)   Axel Thimm
• [Asterisk-Users] Re: Received VNAK: resending outstanding frames?   Carey O'Shea
• [Asterisk-Users] HELP! Bad sound quality   Alex Brett
• [Asterisk-Users] Digium TE110P & TDM400P - always found new hardware on CentOS 4.3   kevin ling
• [Asterisk-Users] Digium TE110P & TDM400P - always found new hardware on CentOS 4.3   Matteo Brancaleoni
• [Asterisk-Users] Digium cards, so disappointing !   Kevin P. Fleming
• [Asterisk-Users] Digium cards, so disappointing !   Steve Underwood
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Steve Underwood
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Remco Barende
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Remco Barende
• [Asterisk-Users] Digium cards, so disappointing !   Begumisa Gerald M
• [Asterisk-Users] Digium cards, so disappointing !   Remco Barende
• [Asterisk-Users] attended transfer issue   Thomas Artner
• [Asterisk-Users] Digium cards, so disappointing !   Steve Underwood
• [Asterisk-Users] Digium cards, so disappointing !   Steve Underwood
• [Asterisk-Users] Digium cards, so disappointing !   Remco Barende
• [Asterisk-Users] Digium cards, so disappointing !   Steve Underwood
• [Asterisk-Users] asterisk voicemail question   Technical Support
• [Asterisk-Users] Re: callerid name inboune from PRI   Steven
• [Asterisk-Users] attended transfer issue   John Novack
• [Asterisk-Users] asterisk voicemail question   Tom Vile
• RES: [Asterisk-Users] attended transfer issue   dovb
• [Asterisk-Users] Digium cards, so disappointing !   Remco Barende
• [Asterisk-Users] asterisk or ser   Yair Hakak
• [Asterisk-Users] attended transfer issue   Thomas Artner
• [Asterisk-Users] Music on hold problem   Daniel Korndorfer
• [Asterisk-Users] Blacklist a callerID   Technical Support
• [Asterisk-Users] CentOS 4.x Asterisk RPMS   alist
• [Asterisk-Users] asterisk voicemail question   Tofik Suleymanov
• [Asterisk-Users] CDR query   Alex Brett
• [Asterisk-Users] FreePBX in Production systems?   Min Hwan Chang
• [Asterisk-Users] Music on hold problem   Alex Brett
• [Asterisk-Users] Music on hold problem   Lacy Moore - Aspendora
• [Asterisk-Users] MixMonitor and filenames   Eric Jacksch
• [Asterisk-Users] placing call with agi   Jon-o Addleman
• [Asterisk-Users] Automatic 3 Way Call   Alexander Chemeris
• [Asterisk-Users] MixMonitor and filenames   Steve Feinstein
• [Asterisk-Users] Dial from php   Wayne
• [Asterisk-Users] FreePBX in Production systems?   Rob Terhaar
• [Asterisk-Users] Re: Digium cards, so disappointing !   Lonnie Abelbeck
• [Asterisk-Users] Cisco 7960 International   Hermann Wecke
• [Asterisk-Users] FreePBX in Production systems?   pdhales at optusnet.com.au
• [Asterisk-Users] Phones that work well through NAT   jennyw
• [Asterisk-Users] Blacklist a callerID   Jay Milk
• [Asterisk-Users] Phones that work well through NAT   Andrew Kohlsmith
• [Asterisk-Users] Digium cards, so disappointing !   Steve Underwood
• [Asterisk-Users] Digium cards, so disappointing !   Craig Guy
• [Asterisk-Users] Phones that work well through NAT   C F
• [Asterisk-Users] Phones that work well through NAT   Ron Senykoff
• [Asterisk-Users] Background music in call   C F
• [Asterisk-Users] change/toggle flash operator panel components   C F
• [Asterisk-Users] How to terminate ringing call before it is answered?   C F
• [Asterisk-Users] voicemail use external smtp server for sending mail   C F
• [Asterisk-Users] How to terminate ringing call before it is answered   C F
• [Asterisk-Users] Caller ID problem   Raju
• [Asterisk-Users] voicemail email-from   Ronald Wiplinger
• [Asterisk-Users] Re: Cisco 7960 International   Shaun
• [Asterisk-Users] Variables   Shaun
• [Asterisk-Users] Background music in call   Dinesh Nair
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, sodisappointing !)   Koopmann, Jan-Peter
• [Asterisk-Users] Variables   Jon Farmer
• [Asterisk-Users] voicemail email-from   KRTorio
• [Asterisk-Users] FreePBX in Production systems?   stoffell
• [Asterisk-Users] voicemail email-from   kevin ling
• [Asterisk-Users] Digium cards, so disappointing !   Michiel van Baak
• [Asterisk-Users] voicemail email-from   Ronald Wiplinger
• [Asterisk-Users] voicemail email-from   Ronald Wiplinger
• [Asterisk-Users] Background music in call   Steve Totaro
• [Asterisk-Users] Digium cards, so disappointing !   Steve Totaro
• [Asterisk-Users] voicemail email-from   Steve Totaro
• [Asterisk-Users] Digium cards, so disappointing !   Michiel van Baak
• [Asterisk-Users] What is Multi-layered-Access control   cmould
• [Fwd: Re: [Asterisk-Users] voicemail email-from]   Steve Totaro
• [Asterisk-Users] Phones that work well through NAT   Chris Mason (Lists)
• [Fwd: Re: [Asterisk-Users] voicemail email-from]   Ronald Wiplinger
• [Asterisk-Users] How do I limit the lenght of a call   John Rich
• [Asterisk-Users] Phones that work well through NAT   Andrew Kohlsmith
• [Asterisk-Users] Digium cards, so disappointing !   Andrew Kohlsmith
• [Asterisk-Users] How do I limit the lenght of a call   Tim Connolly
• [Fwd: Re: [Asterisk-Users] voicemail email-from]   Steve Totaro
• [Asterisk-Users] Re: Cisco 7960 International   Hermann Wecke
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, sodisappointing !)   Olivier Krief
• [Asterisk-Users] Phones that work well through NAT   Bill Gibbs
• [Asterisk-Users] Can Astcc allow dialing phone number more than once   chawki hammoud
• [Asterisk-Users] te110p and interrupts   Paul Hewlett
• [Fwd: Re: [Asterisk-Users] voicemail email-from]   Ronald Wiplinger
• [Asterisk-Users] Performance: Xeon or Opteron?   Olivier Krief
• [Fwd: Re: [Asterisk-Users] voicemail email-from]   Steve Totaro
• [Asterisk-Users] Digium cards, so disappointing !   Steve Totaro
• [Asterisk-Users] Problems with several SIP Providers (one way echo)   Alex Mosburger
• [Asterisk-Users] Problems with several SIP Providers (one way echo)   Roger Schreiter
• [Asterisk-Users] Problems with several SIP Providers (one way echo)   Roger Schreiter
• [Asterisk-Users] Problems with several SIP Providers (one wayecho)   Alex Mosburger
• [Asterisk-Users] Problems with several SIP Providers (one wayecho)   Roger Schreiter
• [Asterisk-Users] FreePBX in Production systems?   Tzafrir Cohen
• [Asterisk-Users] attended transfer issue   Eric "ManxPower" Wieling
• [Asterisk-Users] Can Astcc allow dialing phone number more than once   Darren Wiebe
• [Asterisk-Users] attended transfer issue   John Novack
• [Asterisk-Users] How do I limit the lenght of a call   f6hqz-m at hamwlan.net
• [Asterisk-Users] Outgoing Ringback Indications IAX vs. SIP   Eric "ManxPower" Wieling
• [Asterisk-Users] Re: Re: Cisco 7960 International   Shaun
• [Asterisk-Users] Re: Re: Cisco 7960 International   Tim Robinson
• [Asterisk-Users] Phones that work well through NAT   C F
• [Asterisk-Users] FreePBX in Production systems?   Lacy Moore - Aspendora
• [Asterisk-Users] Re: Trial Version of Asterisk Interface Available   Doug Lytle
• [Asterisk-Users] External voicemail and MWI on internal phone   Reuben Farrelly
• [Asterisk-Users] Flash Key and R Italian Key   Linux Administator
• [Asterisk-Users] Outgoing Ringback Indications IAX vs. SIP   Steve Feinstein
• [Asterisk-Users] Phones that work well through NAT   Christian Stredicke
• [Asterisk-Users] FreePBX in Production systems?   Min Hwan Chang
• [Asterisk-Users] change/toggle flash operator panel components   Nicolás Gudiño
• [Asterisk-Users] Re: Trial Version of Asterisk Interface Available   Vikram Rangnekar
• [Asterisk-Users] attended transfer issue   Eric "ManxPower" Wieling
• [Asterisk-Users] Unicall and Fax   Carlos Chavez
• [Asterisk-Users] Re: Trial Version of Asterisk Interface Available   Peter Bowyer
• [Asterisk-Users] Cisco 7940/7960 SIP 8.2 Freely Downloadable   Alexander Burke
• [Asterisk-Users] Snom 190, Asterisk and Intercom   Ronald Wiplinger
• [Asterisk-Users] Re: Re: Re: Cisco 7960 International   Shaun
• [Asterisk-Users] cdr_pgsql failing to load in head   Chris Stenton
• [Asterisk-Users] Outgoing Ringback Indications IAX vs. SIP   Tim Panton
• [Asterisk-Users] multiple asterisk process ?   stevanus
• [Asterisk-Users] multiple asterisk process ?   Lee Archer
• [Asterisk-Users] Snom 190, Asterisk and Intercom   Peter J Dean
• [Asterisk-Users] Probs with asterisk   Emmanuel LAZARO
• [Asterisk-Users] te110p and interrupts   Begumisa Gerald M
• [Asterisk-Users] te110p and interrupts   Anton Krall
• [Asterisk-Users] te110p and interrupts   stoffell
• [Asterisk-Users] te110p and interrupts   Begumisa Gerald M
• [Asterisk-Users] SIP conections, with RTP not going trough Asterisk   Tiago Stein D`Agostini
• [Asterisk-Users] cdr_pgsql failing to load in head   Rich Adamson
• [Asterisk-Users] SIP conections, with RTP not going trough Asterisk   Peter Bowyer
• [Asterisk-Users] SIP conections, with RTP not going trough Asterisk   Ronald Wiplinger
• [Asterisk-Users] SIP conections, with RTP not going trough Asterisk   Rich Adamson
• [Asterisk-Users] Probs with asterisk   Alex Mosburger
• [Asterisk-Users] SIP conections, with RTP not going trough Asterisk   Tiago Stein D`Agostini
• [Asterisk-Users] SIP conections, with RTP not going trough Asterisk   Alex Mosburger
• [Asterisk-Users] SIP conections, with RTP not going trough Asterisk   Peter Bowyer
• [Asterisk-Users] SIP conections, with RTP not going trough Asterisk   Tiago Stein D`Agostini
• [Asterisk-Users] Probs with asterisk   John covici
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Rich Adamson
• [Asterisk-Users] Asterisk redundancy   Joseph Rothstein
• [Asterisk-Users] SIP conections, with RTP not going trough Asterisk   stoffell
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Andrew Kohlsmith
• [Asterisk-Users] Probs with asterisk   Emmanuel LAZARO
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Lee Howard
• [Asterisk-Users] Probs with asterisk   Emmanuel LAZARO
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Andrew Kohlsmith
• [Asterisk-Users] Asterisk redundancy   Julian Lyndon-Smith
• [Asterisk-Users] Asterisk redundancy   Tim Panton
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Nicholas Kathmann
• [Asterisk-Users] CentOS 4.x Asterisk RPMS   Erick Perez
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Lee Howard
• [Asterisk-Users] CDR query   Alex Brett
• [Asterisk-Users] Agents, Queues, and Voicemail   Kyle Sexton
• [Asterisk-Users] multiple asterisk process ?   Moises Silva
• [Asterisk-Users] Unicall and Fax   Paulo Scardine
• [Asterisk-Users] Agents, Queues, and Voicemail   Johann
• [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade   Jim Rice
• [Asterisk-Users] Agents, Queues, and Voicemail   Kyle Sexton
• [Asterisk-Users] asterisk voicemail question   Don Pobanz
• [Asterisk-Users] Asterisk redundancy   Aaron Daniel
• [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade   Andrew Kohlsmith
• [Asterisk-Users] Don't see my post   John Rich
• [Asterisk-Users] CAPI Installation Eicon Diva Server   Armin Schindler
• [Asterisk-Users] Don't see my post   Jean-Michel Hiver
• [Asterisk-Users] Quick question   Tomislav Parčina
• [Asterisk-Users] Quick question   Alberto Sagredo
• [Asterisk-Users] Don't see my post   TWV
• [Asterisk-Users] voicemail use external smtp server for sendingmail   Steve Jones
• [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade   Jonathan k. Creasy
• [Asterisk-Users] Don't see my post   John Rich
• [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser   TWV
• [Asterisk-Users] IAX phone hardware recommendation   John Rich
• [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser   Dr. Michael J. Chudobiak
• [Asterisk-Users] Setting CDR dnid and Billing   Douglas Garstang
• [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade   Jim Rice
• [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser   TWV
• [Asterisk-Users] cdr_pgsql failing to load in head   Brian Capouch
• [Asterisk-Users] multiple asterisk process ?   Paul Hewlett
• [Asterisk-Users] Billing Server Open Source   broadbandvoice at comcast.net
• [Asterisk-Users] Asterisk Like Phone Switch ?   Luz Lopez
• [Asterisk-Users] multiple asterisk process ?   Dave Cotton
• [Asterisk-Users] te110p and interrupts   Paul Hewlett
• [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade   Andrew Kohlsmith
• [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording   Henri Herscher
• [Asterisk-Users] Sip Notify cisco-check-cfg - Does it still work with 8.2?   Brent Torrenga
• [Asterisk-Users] multiple asterisk process ?   Moises Silva
• [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser   Andrew Latham
• [Asterisk-Users] Billing Server Open Source   William Piper
• [Asterisk-Users] Billing Server Open Source   William Piper
• RES: [Asterisk-Users] attended transfer issue   Kevin Bockman
• [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade   Jim Rice
• [Asterisk-Users] codec negotiation   Ronald Wiplinger
• [Asterisk-Users] H.323 question, take so long time to call   welemon lee
• [Asterisk-Users] astcc and inwards billing   Ronald Wiplinger
• [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade   Jim Rice
• [Asterisk-Users] Asterisk hardware for new office suggestion   Simone
• [Asterisk-Users] Cannot dial out with Polycom 501 after upgrade   Andrew Kohlsmith
• [Asterisk-Users] Orative   Dean Collins
• [Asterisk-Users] Asterisk hardware for new office suggestion   stoffell
• [Asterisk-Users] Orative   Douglas Garstang
• [Asterisk-Users] Orative   Dean Collins
• [Asterisk-Users] Orative   Andrew D Kirch
• [Asterisk-Users] Orative   wendell hamilton
• [Asterisk-Users] CentOS 4.x Asterisk RPMS   alist
• [Asterisk-Users] astcc and inwards billing   JP Carballo
• [Asterisk-Users] Orative   Brian Capouch
• [Asterisk-Users] CAPI Installation Eicon Diva Server   Avi Miller
• [Asterisk-Users] Orative   Douglas Garstang
• [Asterisk-Users] Asterisk hyperthreading compiling.   Wai Wu
• [Asterisk-Users] Orative   Andrew Latham
• [Asterisk-Users] Asterisk hyperthreading compiling.   Justin Tunney
• [Asterisk-Users] Asterisk hyperthreading compiling.   Matt Roth
• [Asterisk-Users] Asterisk hyperthreading compiling.   Matt Florell
• [Asterisk-Users] Asterisk settings for roaming users   Andy Tan
• [Asterisk-Users] Looking for a good VoIP Provider in the UK-   Maxx Lobo
• [Asterisk-Users] Sending SIP NOTIFY   TWV
• [Asterisk-Users] Looking for a good VoIP Provider in the UK-   brett at websmyths.com
• [Asterisk-Users] Looking for a good VoIP Provider in the UK-   George Pajari
• [Asterisk-Users] Orative   Andrew D Kirch
• [Asterisk-Users] Sending SIP NOTIFY   TWV
• [Asterisk-Users] Cisco 7940/7960 SIP 8.2 Freely Downloadable   Hermann Wecke
• [Asterisk-Users] Variables   pdhales at optusnet.com.au
• [Asterisk-Users] SIP TAPI   Jerry Jones
• [Asterisk-Users] DTMF Tones in RTP Payload as Well as in Events = Duplicate Tones   Wolfgang Borgon
• [Asterisk-Users] Looking for a good VoIP Provider in the UK-   Steve Kennedy
• [Asterisk-Users] voicemail use external smtp server for sendingmail   C F
• [Asterisk-Users] Asterisk hardware for new office suggestion   kevin ling
• [Asterisk-Users] multiple asterisk process ?   stevanus
• [Asterisk-Users] voicemail use external smtp server for sendingmail   Jerry Workman
• [Asterisk-Users] te110p and interrupts   Anton Krall
• [Asterisk-Users] Looking for a good VoIP Provider in the UK-   Peter Bowyer
• [Asterisk-Users] Asterisk redundancy   Olivier Krief
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Olivier Krief
• [Asterisk-Users] Quick question   Tomislav Parčina
• [Asterisk-Users] Asterisk hardware for new office suggestion   Kevin Bockman
• [Asterisk-Users] Sip.conf   Tomislav Parčina
• [Asterisk-Users] multiple asterisk process ?   Tim Panton
• [Asterisk-Users] Quick question   Jeremy McNamara
• [Asterisk-Users] multiple asterisk process ?   Dave Cotton
• [Asterisk-Users] multiple asterisk process ?   Lee Archer
• [Asterisk-Users] multiple asterisk process ?   Dave Cotton
• [Asterisk-Users] voicemail use external smtp server for sendingmail   picciuX
• [Asterisk-Users] multiple asterisk process ?   Lee Archer
• [Asterisk-Users] multiple asterisk process ?   Dave Cotton
• [Asterisk-Users] multiple asterisk process ?   stevanus
• [Asterisk-Users] HardPhone PlanetVIP-150T - Starts music on Hold and i can't get the call again   Marco Mouta
• [Asterisk-Users] Re: HardPhone PlanetVIP-150T - Starts music on Hold and i can't get the call again   Marco Mouta
• [Asterisk-Users] multiple asterisk process ?   Lee Archer
• [Asterisk-Users] Sip.conf   kevin ling
• [Asterisk-Users] Grandstream Budgetone and Mac mini?   Dmitry Ivanov
• [Asterisk-Users] Sip.conf   Ivan Meic
• [Asterisk-Users] Sip.conf   Alejandro Vargas
• [Asterisk-Users] RE: Sip.conf   Tomislav Parčina
• [Asterisk-Users] Re: Sip.conf   Tomislav Parčina
• [Asterisk-Users] Re: Quick question   Tomislav Parčina
• [Asterisk-Users] Re: Quick question   Tomislav Parčina
• [Asterisk-Users] Cisco 7970 SIP - few questions   Tomislav Parčina
• [Asterisk-Users] multiple asterisk process ?   Tim Panton
• [Asterisk-Users] Using ISDN MSNs for dialing out of Asterisk   Christian Gröger
• [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud   broadbandvoice at comcast.net
• [Asterisk-Users] Granstream GXP2000 Distinctive tones   Cory Hawkless
• [Asterisk-Users] Asterisk/FreePBX/Alcatel2400   Andy Green
• [Asterisk-Users] incoming calls dropout on PRI over TE110p   broadbandvoice at comcast.net
• [Asterisk-Users] Granstream GXP2000 Distinctive tones   Gareth Blades
• [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud   Gareth Blades
• [Asterisk-Users] Grandstream Budgetone and Mac mini?   Mark Phillips
• [Asterisk-Users] Change email/pager VM alerts body text dynamically?   jimw at midwest-connections.com
• [Asterisk-Users] rpms updated to 1.2.7.1 (was: Asterisk 1.2.7.1Released)   Mimmus
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Nicholas Kathmann
• [Asterisk-Users] Re: rpms updated to 1.2.7.1 (was: Asterisk 1.2.7.1Released)   Axel Thimm
• [Asterisk-Users] Grandstream Budgetone and Mac mini?   Rusty Dekema
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Doug Lytle
• [Asterisk-Users] voicemail kicking in after user has already disconnected   Mike Garey
• [Asterisk-Users] Grandstream Budgetone and Mac mini?   Jerry Jones
• [Asterisk-Users] Noise on IAX or SIP trunk between 2 Asterisk   kritikus Araklidas
• [Asterisk-Users] voicemail kicking in after user has already disconnected   Jerry Jones
• [Asterisk-Users] Re: Cisco 7940/7960 SIP 8.2 Freely   Brent Torrenga
• [Asterisk-Users] Phones that work well through NAT   Sean Garland
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)   Rich Adamson
• [Asterisk-Users] Grandstream Budgetone and Mac mini?   Tomislav Vojvodic
• [Asterisk-Users] IVR: playing multiple streams simultaneously?   Herchi Silviu
• [Asterisk-Users] Phones that work well through NAT   Andrew Kohlsmith
• [Asterisk-Users] Polycom Microbrowser   hgaillac-sip at yahoo.fr
• [Asterisk-Users] rpms updated to 1.2.7.1 (was: Asterisk 1.2.7.1Released)   Patrick
• [Asterisk-Users] CAPI Installation Eicon Diva Server   nkohl at maninvestments.com
• [Asterisk-Users] bad voice quality   Dumpolid Exeplish
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Steve Underwood
• [Asterisk-Users] IVR and voicemail issues ?   TAG
• [Asterisk-Users] Asterisk code help   Matt Riddell [IT]
• [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser   Hirosh Dabui
• [Asterisk-Users] bad voice quality   Alex Mosburger
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Doug Lytle
• [Asterisk-Users] correct version of asterisk for oh323   yusuf
• [Asterisk-Users] Grandstream Budgetone and Mac mini?   The VoIP Connection
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Lee Howard
• [Asterisk-Users] Asterisk Performance 350 Concurrent Channels Working Nicely   JR Richardson
• [Asterisk-Users] correct version of asterisk for oh323   Herchi Silviu
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Olivier Krief
• [Asterisk-Users] Voicemail problem   Daniel Korndorfer
• [Asterisk-Users] Help Getting Local Exchange Dialtone on PRI   Christoph Adomeit
• [Asterisk-Users] correct version of asterisk for oh323   yusuf
• [Asterisk-Users] Performance: Xeon or Opteron?   yusuf
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Nicholas Kathmann
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Doug Lytle
• [Asterisk-Users] BSR 1000 and Asterisk   Carlos Alberto Bernat Orozco
• [Asterisk-Users] Gizmo Call In   Toke
• [Asterisk-Users] Grandstream Budgetone and Mac mini?   Nathan Bowyer
• [Asterisk-Users] eyeBeam + ASterisk 1.2.7.1 + Instant Message   João Paulo Antunes
• [Asterisk-Users] Asterisk Performance 350 Concurrent ChannelsWorking Nicely   Douglas Garstang
• [Asterisk-Users] eyeBeam + ASterisk 1.2.7.1 + Instant Message   Joshua Colp
• [Asterisk-Users] IVR: playing multiple streams simultaneously?   Alexander Lopez
• [Asterisk-Users] Voicemail Issue - Failed to lock path   Brent Torrenga
• [Asterisk-Users] correct version of asterisk for oh323   ADEGOKE ARUNA
• [Asterisk-Users] Asterisk crash with Digium   Rudolf E. Steiner
• [Asterisk-Users] Double Ring - TelIAX/Cisco 79[46]0   Brent Torrenga
• [Asterisk-Users] ISDN in Japan?   Chris Earle (CBL)
• [Asterisk-Users] T1 to cross connect remote PBX and asterisk   Damon Estep
• [Asterisk-Users] Polycom Microbrowser   Mojo with Horan & Company, LLC
• [Asterisk-Users] ISDN in Japan?   Armin Schindler
• [Asterisk-Users] eyeBeam + ASterisk 1.2.7.1 + Instant Message   Douglas Garstang
• [Asterisk-Users] Aastra 9133i Phones Asterisk 1.2.6 and MWI   Matt
• [Asterisk-Users] Asterisk Performance 350 Concurrent   JR Richardson
• [Asterisk-Users] PRI blocking on incoming calls   Kevin Savoy
• [Asterisk-Users] T1 to cross connect remote PBX and asterisk   C F
• [Asterisk-Users] T1 to cross connect remote PBX and asterisk   Damon Estep
• [Asterisk-Users] Dialplan - strip IDD prefix and insert another   broadbandvoice at comcast.net
• [Asterisk-Users] re: Sixtel Services   broadbandvoice at comcast.net
• [Asterisk-Users] Realtime goto problem   Pedro Nunes
• [Asterisk-Users] re: Sixtel Services   broadbandvoice at comcast.net
• [Asterisk-Users] PRI blocking on incoming calls   Oscar Carriles
• [Asterisk-Users] Asterisk redundancy   Benjamin Lawetz
• [Asterisk-Users] T1 to cross connect remote PBX and asterisk   Jim Houser
• [Asterisk-Users] PRI blocking on incoming calls   Kevin Savoy
• [Asterisk-Users] Receiving Faxes...   Christian Gröger
• [Asterisk-Users] Change email/pager VM alerts body text dynamically?   Benjamin Lawetz
• [Asterisk-Users] Re: HardPhone PlanetVIP-150T - Starts music on Hold and i can't get the call again   Andrew D Kirch
• [Asterisk-Users] Warning message   Dov Bigio
• [Asterisk-Users] T1 to cross connect remote PBX and asterisk   Doug Lytle
• [Asterisk-Users] Warning message   Kevin P. Fleming
• [Asterisk-Users] PRI blocking on incoming calls   Oscar Carriles
• [Asterisk-Users] FW: NuFone Update: DIDs   Wes Baehr
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 20, Issue 31   broadbandvoice at comcast.net
• [Asterisk-Users] re: Sixtel Services   Steve Jones
• [Asterisk-Users] T1 to cross connect remote PBX and asterisk   Damon Estep
• [Asterisk-Users] T1 to cross connect remote PBX and asterisk   Jim Houser
• [Asterisk-Users] FW: NuFone Update: DIDs   Seth Remington
• [Asterisk-Users] ISDN in Japan?   Chris Earle (CBL)
• [Asterisk-Users] FW: NuFone Update: DIDs   Tom Vile
• [Asterisk-Users] FW: NuFone Update: DIDs   Darrick Hartman
• [Asterisk-Users] UK Asterisk sound files   Steve Kennedy
• [Asterisk-Users] FW: NuFone Update: DIDs   Brian Capouch
• [Asterisk-Users] polycom blind transfer button   Anton Krall
• [Asterisk-Users] Asterisk & GNUDialer issue   Facundo Ameal
• [Asterisk-Users] Re: UK Asterisk sound files   Tony Mountifield
• [Asterisk-Users] Polycom IP 501 buddy list: Got SIP response 500 "Internal Server Error"   Mike Garey
• [Asterisk-Users] polycom blind transfer button   Jonathan k. Creasy
• [Asterisk-Users] Problem Using Asterisk Call Files with Zap PRI   Adam Robins
• [Asterisk-Users] Outbound calls are failing   nrbwpi at gmail.com
• [Asterisk-Users] T1 to cross connect remote PBX and asterisk   C F
• [Asterisk-Users] Re: UK Asterisk sound files   Steve Kennedy
• [Asterisk-Users] Dialplan - strip IDD prefix and insert another   AR Tarzi
• [Asterisk-Users] ISDN in Japan?   Andrew Latham
• [Asterisk-Users] ISDN in Japan?   Jason Frisch
• [Asterisk-Users] ISDN in Japan?   Andrew Latham
• [Asterisk-Users] ISDN in Japan?   Jason Frisch
• [Asterisk-Users] How To - Building a VoIP-PSTN Gateway with Asterisk   Jean-Louis curty
• [Asterisk-Users] UK Asterisk sound files   Tzafrir Cohen
• [Asterisk-Users] Outgoing voice distortion with Unicall   Carlos Chavez
• [Asterisk-Users] UK Asterisk sound files   Steve Kennedy
• [Asterisk-Users] T1 to cross connect remote PBX and asterisk   Damon Estep
• [Asterisk-Users] Remember the incoming context?   Edwin Groothuis
• [Asterisk-Users] Sending SIP NOTIFY   Mojo with Horan & Company, LLC
• [Asterisk-Users] Asterisk & GNUDialer issue   Josué Conti
• [Asterisk-Users] Remember the incoming context?   Christian Gröger
• [Asterisk-Users] Voicemail exits   Daniel Korndorfer
• [Asterisk-Users] Outgoing voice distortion with Unicall   Moises Silva
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Anton Krall
• [Asterisk-Users] Performance: Xeon or Opteron?   Anton Krall
• [Asterisk-Users] polycom blind transfer button   Anton Krall
• [Asterisk-Users] Phones that work well through NAT   Sean Garland
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Steve Underwood
• [Asterisk-Users] Re: Variables   Shaun
• [Asterisk-Users] Don't see my post   billy at kersting.com
• [Asterisk-Users] Re: Variables   Paul Hales
• [Asterisk-Users] Asterisk service crashes   billy at kersting.com
• [Asterisk-Users] Outgoing voice distortion with Unicall   Carlos Chavez
• [Asterisk-Users] Remember the incoming context?   Edwin Groothuis
• [Asterisk-Users] Outgoing voice distortion with Unicall   CC Jay
• [Asterisk-Users] Welltech Welgate 3804 FXO Configs   Ronald Wiplinger
• [Asterisk-Users] Sending SIP NOTIFY / How to get remote SIP port?   TWV
• [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser   TWV
• [Asterisk-Users] Polycom 501 resource full problems ...   Anthony Rodgers
• [Asterisk-Users] Asterisk redundancy   Gary Richardson
• [Asterisk-Users] Outbound calls are failing   Time Bandit
• [Asterisk-Users] Double Ring - TelIAX/Cisco 79[46]0   Jonathan Feally
• [Asterisk-Users] Outgoing voice distortion with Unicall   Carlos Chavez
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Olivier Krief
• [Asterisk-Users] Outgoing voice distortion with Unicall   Stepan Hradsky
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Anton Krall
• [Asterisk-Users] Outgoing voice distortion with Unicall   Anton Krall
• [Asterisk-Users] T1 Newbie Question about IDA-P   Sam Tam
• [Asterisk-Users] need stand-alone FXO ports   Ronald Wiplinger
• [Asterisk-Users] Re: Re: Variables   Shaun
• [Asterisk-Users] need stand-alone FXO ports   Kerry Garrison
• [Asterisk-Users] Re: Re: Variables   Shaun
• [Asterisk-Users] correct version of asterisk for oh323   Herchi Silviu
• [Asterisk-Users] IVR: playing multiple streams simultaneously?   Giridhar Reddy Bandi
• [Asterisk-Users] Sending SIP NOTIFY / How to get remote SIP port?   Freddi Hansen
• [Asterisk-Users] IVR: playing multiple streams simultaneously?   Herchi Silviu
• [Asterisk-Users] Remember the incoming context?   Herchi Silviu
• [Asterisk-Users] Voicemail problem   Giridhar Reddy Bandi
• [Asterisk-Users] Receiving Faxes...   Herchi Silviu
• [Asterisk-Users] Background music in call   nzrh
• [Asterisk-Users] LCDC and lcd.conf, p_, c_   Ronald Wiplinger
• [Asterisk-Users] Re: Remember the incoming context?   Tony Mountifield
• [Asterisk-Users] Sending SIP NOTIFY / How to get remote SIP port?   TWV
• [Asterisk-Users] Receiving Faxes...   Gareth Blades
• [Asterisk-Users] Asterisk service crashes   Giridhar Reddy Bandi
• [Asterisk-Users] voicemail use external smtp server for sendingmail   nik600
• [Asterisk-Users] Asterisk service crashes   Gareth Blades
• [Asterisk-Users] Grandstream Budgetone and Mac mini?   Dmitry Ivanov
• [Asterisk-Users] SIP transfers of queued calls doesn't make agent available   Mark Roeten
• [Asterisk-Users] Re: Re: Variables   Shaun
• [Asterisk-Users] polycom unable to start recoding   Giridhar Reddy Bandi
• [Asterisk-Users] SIP transfers of queued calls doesn't make agent available   picciuX
• [Asterisk-Users] Snom 360 Firmware 6.0 - Microbrowser   Hirosh Dabui
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Doug Lytle
• [Asterisk-Users] Calls stuck in queue...   Shaun
• [Asterisk-Users] SIP transfers of queued calls doesn't make agentavailable   Mark Roeten
• [Asterisk-Users] error when executing sphinx!!!   serge messa
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent and called party stills MOH   Marco Mouta
• [Asterisk-Users] eyeBeam + ASterisk 1.2.7.1 + Instant Message   hgaillac-sip at yahoo.fr
• [Asterisk-Users] FW: NuFone Update: DIDs   Chris Mason (Lists)
• [Asterisk-Users] Asterisk and 7960s   dave
• [Asterisk-Users] Asterisk service crashes   Josué Conti
• [Asterisk-Users] Asterisk 1.2.7.1 and IAX modem / channel   Pimjai Wesnarat
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Doug Lytle
• [Asterisk-Users] Asterisk 1.2.7.1 and IAX modem / channel   Doug Lytle
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Marco Mouta
• [Asterisk-Users] Asterisk 1.2.7.1 and IAX modem / channel   Michael Strelnikov
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Doug Lytle
• [Asterisk-Users] Asterisk 1.2.7.1 and IAX modem / channel   Pimjai Wesnarat
• [Asterisk-Users] Billing Server Open Source   Mindaugas Kezys
• [Asterisk-Users] Re: new_callback_call and conf disconnect   Abhimanyu Rapria
• [Asterisk-Users] Unable to allocate socket: Too may open files   Stefan Günther
• [Asterisk-Users] Problem with Voice quality, please help   mkumar at mantragroup.com
• [Asterisk-Users] oh323: asterisk crashes on a dial   yusuf
• [Asterisk-Users] Unable to allocate socket: Too may open files   Oscar Carriles
• [Asterisk-Users] FW: NuFone Update: DIDs (Correction)   Wes Baehr
• [Asterisk-Users] Asterisk service crashes   William Piper
• [Asterisk-Users] Asterisk service crashes   Steve Totaro
• [Asterisk-Users] PRI caller ID   Jonathan k. Creasy
• [Asterisk-Users] FW: NuFone Update: DIDs (Correction)   Brian C. Fertig
• [Asterisk-Users] PRI caller ID   Jerry Jones
• [Asterisk-Users] Asterisk service crashes   Josué Conti
• [Asterisk-Users] ISDN in Japan?   isamar at isamarmaia.org
• [Asterisk-Users] Asterisk service crashes   Steve Totaro
• [Asterisk-Users] PRI caller ID   Jonathan k. Creasy
• [Asterisk-Users] Using Asterisk modules in external application   Leonardo (listas)
• [Asterisk-Users] Re: Asterisk and 7960s   Brent Torrenga
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Marco Mouta
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Gareth Blades
• [Asterisk-Users] Kernel panic - suggestions?   Rich Adamson
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Marco Mouta
• [Asterisk-Users] Call Center with No TDM components   Abhimanyu Rapria
• [Asterisk-Users] Re: Asterisk and 7960s   dave
• [Asterisk-Users] Kernel panic - suggestions?   Kristian Kielhofner
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Doug Lytle
• [Asterisk-Users] Kernel panic - suggestions?   Damon Estep
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Doug Lytle
• [Asterisk-Users] Kernel panic - suggestions?   Doug Lytle
• [Asterisk-Users] Kernel panic - suggestions?   Moises Silva
• [Asterisk-Users] FW: NuFone Update: DIDs   Wilson Pickett
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Marco Mouta
• [Asterisk-Users] Kernel panic - suggestions?   Rich Adamson
• [Asterisk-Users] Problems with several SIP Providers (one wayecho)   Eric "ManxPower" Wieling
• [Asterisk-Users] attended transfer issue   Eric "ManxPower" Wieling
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Alexander Lopez
• [Asterisk-Users] Re: Asterisk and 7960s   Brent Torrenga
• [Asterisk-Users] Sip channel variables   Dov Bigio
• [Asterisk-Users] PRI caller ID   Mimmus
• [Asterisk-Users] Re: Asterisk and 7960s   Aaron Daniel
• [Asterisk-Users] Phones that work well through NAT   Andrew Kohlsmith
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Johann Steinwendtner
• [Asterisk-Users] Sip channel variables   VladK
• [Asterisk-Users] Call Center with No TDM components   Begumisa Gerald M
• [Asterisk-Users] Re: Asterisk and 7960s   Michiel van Baak
• [Asterisk-Users] Outgoing voice distortion with Unicall   Carlos Chavez
• [Asterisk-Users] Asterisk service crashes   Ken Godee
• [Asterisk-Users] Callerid matching in extensions.conf   Douglas Garstang
• [Asterisk-Users] SLIN format   Steve Kennedy
• [Asterisk-Users] Re: Asterisk and 7960s   dave
• [Asterisk-Users] Asterisk service crashes   William Piper
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Doug Lytle
• [Asterisk-Users] Outgoing voice distortion with Unicall   Rich Adamson
• [Asterisk-Users] Delayed voice for 10 secs   Carlos Alberto Bernat Orozco
• [Asterisk-Users] SLIN format   Kevin P. Fleming
• [Asterisk-Users] Asterisk 1.2.6 and 9133i   Matt
• [Asterisk-Users] What's the best way to combine multiple VOIP lines into a single number?   Leo Burd
• [Asterisk-Users] Unable to allocate socket: Too may open files   Matt Roth
• [Asterisk-Users] Callerid matching in extensions.conf   Michiel van Baak
• [Asterisk-Users] What's the best way to combine multiple VOIP lines into a single number?   Andrew Kohlsmith
• [Asterisk-Users] bad voice quality   Dumpolid Exeplish
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Lee Howard
• [Asterisk-Users] How to route all incoming calls on an analog trunk to a specific ring group   Tony Maupin
• [Asterisk-Users] Re: SLIN format   Tony Mountifield
• [Asterisk-Users] Asterisk service crashes   William Piper
• [Asterisk-Users] Re: SLIN format   Watkins, Bradley
• [Asterisk-Users] Where to buy Eicon DIVA cards   Klaus Darilion
• [Asterisk-Users] attended transfer issue   Damon Estep
• [Asterisk-Users] clearing "stuck" channels without a restart   Bill Gibbs
• [Asterisk-Users] attended transfer issue   Andrew Kohlsmith
• [Asterisk-Users] Re: Asterisk and 7960s   Daryl Jones
• [Asterisk-Users] Call Center with No TDM components   Abhimanyu Rapria
• [Asterisk-Users] Meetme codec translation and callerID library.   tom
• [Asterisk-Users] Outgoing voice distortion with Unicall   Carlos Chavez
• [Asterisk-Users] Re: Asterisk and 7960s   Aaron Daniel
• [Asterisk-Users] Asterisk IVR / Scalability   Giridhar Reddy Bandi
• [Asterisk-Users] clearing "stuck" channels without a restart   Christopher Mayfield
• [Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly   Dave Fullerton
• [Asterisk-Users] Asterisk 1.2.7.1 and IAX modem / channel   Lee Howard
• [Asterisk-Users] clearing "stuck" channels without a restart   Bill Gibbs
• [Asterisk-Users] PRI blocking on incoming calls   Kevin Savoy
• [Asterisk-Users] Re: regexp in gotoif   Christian B
• [Asterisk-Users] clearing "stuck" channels without a restart   Christopher Mayfield
• [Asterisk-Users] Re: SLIN format   Tony Mountifield
• [Asterisk-Users] Ring a grop of extension, then playback a file, then transfer to external number   Andre Courchesne - Consultant
• [Asterisk-Users] Ring a grop of extension, then playback a file, then transfer to external number   Jason Adams
• [Asterisk-Users] Fwd: sip.conf and jump from register to the extension   Thomas Winter
• [Asterisk-Users] RE: Delayed voice for 10 secs   Cavanna, Richard
• [Asterisk-Users] Fwd: sip.conf and jump from register to the extension   Aaron Daniel
• [Asterisk-Users] Fwd: sip.conf and jump from register to the extension   Thomas Winter
• [Asterisk-Users] Fwd: sip.conf and jump from register to the extension   Aaron Daniel
• [Asterisk-Users] Voice mail issuse when pressing 0   Doug Lytle
• [Asterisk-Users] Fwd: sip.conf and jump from register to the extension   Thomas Winter
• [Asterisk-Users] Fwd: sip.conf and jump from register to the extension   Aaron Daniel
• [Asterisk-Users] Error installing asterisk   Luis Fernando Ramírez Cueva
• [Asterisk-Users] Codec problem from SIP to H323   Alejandro Mejía Evertsz
• [Asterisk-Users] Error installing asterisk   Jesus Mogollon
• [Asterisk-Users] Codec problem from SIP to H323   Oliver Vermeulen
• [Asterisk-Users] Codec problem from SIP to H323   Alejandro Mejía Evertsz
• [Asterisk-Users] attended transfer issue   Damon Estep
• [Asterisk-Users] What's the best way to combine multiple VOIP lines into a single number?   tom
• [Asterisk-Users] What's the best way to combine multiple VOIP lines into a single number?   Tim Litwiller
• [Asterisk-Users] Fwd: sip.conf and jump from register to the extension   tom
• [Asterisk-Users] MultiBRI in Australia - found one - maybe   Hadley Rich
• [Asterisk-Users] attended transfer issue   Eric "ManxPower" Wieling
• [Asterisk-Users] Ring a grop of extension, then playback a file, then transfer to external number   Eric "ManxPower" Wieling
• [Asterisk-Users] Ring a grop of extension, then playback a file, then transfer to external number   Alexander Lopez
• [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1   billy at kersting.com
• [Asterisk-Users] Meetme codec translation and callerID library.   Alexander Lopez
• [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1   Alexander Lopez
• [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1   billy at kersting.com
• [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1   Alexander Lopez
• [Asterisk-Users] lost audio after zaptel   Doug Langley
• [Asterisk-Users] lost audio after zaptel   Paul Hales
• [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1   billy at kersting.com
• [Asterisk-Users] Asterisk service crashes   Steve Jones
• [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1   Alexander Lopez
• [Asterisk-Users] attended transfer issue   John Novack (port)
• [Asterisk-Users] Upgrade from 1.2.4 to 1.2.7.1   billy at kersting.com
• [Asterisk-Users] dundi trouble   陈帆
• [Asterisk-Users] Asterisk service crashes   billy at kersting.com
• [Asterisk-Users] What's the best way to combine multiple VOIP lines into a single number?   Andrew Kohlsmith
• [Asterisk-Users] dundi trouble   Ronald Wiplinger
• [Asterisk-Users] ANNOUNCE: Asterisk Jobs and Consulting Site   Matt Gibson
• [Asterisk-Users] sip.conf codecs: ulaw, alaw and g729   J Shaun Hofer
• [Asterisk-Users] TDMoE   kritikus Araklidas
• [Asterisk-Users] ANNOUNCE: Asterisk Jobs and Consulting Site   Andrew Kohlsmith
• [Asterisk-Users] Meetme codec translation and callerID library.   Jay Milk
• [Asterisk-Users] asterisk 1.2.7.1 crashing my newly built system   T.S
• [Asterisk-Users] dundi trouble   陈帆
• [Asterisk-Users] dundi trouble   陈帆
• [Asterisk-Users] Jingle support - can we test the feature ?   Robert Rozman
• [Asterisk-Users] Dial two extensions at the SAME time and connect them when possible   Robert Rozman
• [Asterisk-Users] How to stop Asterisk from picking up my phone?   Linuxnizer The Mesmorizer
• [Asterisk-Users] Where to buy Eicon DIVA cards   David Waugh
• [Asterisk-Users] How to stop Asterisk picking up my incoming calls?   Linuxnizer The Mesmorizer
• [Asterisk-Users] Happy story   Linuxnizer The Mesmorizer
• [Asterisk-Users] Re: Remember the incoming context?   Benny Amorsen
• [Asterisk-Users] Meetme codec translation and callerID library.   tom
• [Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly   Vahan Yerkanian
• [Asterisk-Users] Fwd: sip.conf and jump from register to the extension   Thomas Winter
• Faxing and PCI (was Re: [Asterisk-Users] Digium cards,   Olivier Krief
• [Asterisk-Users] asterisk and siemens hipath 3500   Ricardo
• [Asterisk-Users] avm b1with chan capi and siemens hipath   Ricardo
• [Asterisk-Users] avm b1with chan capi and siemens hipath   Armin Schindler
• [Asterisk-Users] why DUNDi ${IPADDR} can not transfer to 127.0.0.1?   陈帆
• [Asterisk-Users] SPA-3000 Bug? Dropped calls while registering.   Dana Harding
• [Asterisk-Users] Asterisk & MGCP reinvite   Nicolas TOUSSAINT
• [Asterisk-Users] asterisk + mobicents   hgaillac-sip at yahoo.fr
• [Asterisk-Users] still some moh troubles   Bart van Daal
• [Asterisk-Users] transforming g729 files to wav files   Christian Wengel
• [Asterisk-Users] Best Fax send through Asterisk plan?   hoowa sun
• [Asterisk-Users] Fwd: why DUNDi ${IPADDR} has been transfered to 127.0.0.1?   陈帆
• [Asterisk-Users] asterisk + mobicents   Stefan Reuter
• [Asterisk-Users] ANNOUNCE: Asterisk Jobs and Consulting Site   Steve Totaro
• [Asterisk-Users] Playback(something,noanswer) on Zap?   Dmitry Ivanov
• [Asterisk-Users] CDRs and billing   Joao Pereira
• [Asterisk-Users] aterisk+h323 trunk?!   Tic Pavlin
• [Asterisk-Users] attended transfer issue   Olivier Krief
• [Asterisk-Users] Using ISDN MSNs for dialing out of Asterisk   Dmitry Ivanov
• [Asterisk-Users] CDRs and billing   Chris Mason (Lists)
• [Asterisk-Users] Internet connection   Giordano Grandis
• [Asterisk-Users] still some moh troubles   Doug Lytle
• [Asterisk-Users] SPA-3000 Bug? Dropped calls while registering.   Moises Silva
• [Asterisk-Users] still some moh troubles   Bart van Daal
• [Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly   Bryan Boatright
• [Asterisk-Users] ANNOUNCE: Asterisk Jobs and Consulting Site   Bart Fisher
• [Asterisk-Users] aterisk+h323 trunk?!   yusuf
• [Asterisk-Users] Asterisk & MGCP reinvite   Alejandro Vargas
• [Asterisk-Users] Ring a grop of extension, then playback a file, then transfer to external number   Don Pobanz
• [Asterisk-Users] How to stop Asterisk picking up my incoming calls?   Time Bandit
• [Asterisk-Users] avm b1with chan capi and siemens hipath   Ricardo
• [Asterisk-Users] Asterisk service crashes   William Piper
• [Asterisk-Users] Jingle support - can we test the feature ?   Time Bandit
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 21, Issue 113   Carlos Alberto Bernat Orozco
• [Asterisk-Users] CDRs and billing   Joao Pereira
• [Asterisk-Users] CDRs and billing   Alejandro Mejía Evertsz
• [Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly   Steven Ringwald
• [Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly   Dave Fullerton
• [Asterisk-Users] Asterisk service crashes   William Piper
• [Asterisk-Users] Cubix Softphone + Asterisk 1.2.6   Peter Beckman
• [Asterisk-Users] does anyone know anything about chan_btp or btpd?   Mike Garey
• [Asterisk-Users] Jingle support - can we test the feature ?   Robert Rozman
• [Asterisk-Users] zaptel and zapata configuration   Mongi LASSOUED
• [Asterisk-Users] MeetMe: lots of buffer overruns/underruns when connecting over IAX   Erik Hensema
• [Asterisk-Users] Playback(something,noanswer) on Zap?   Kevin P. Fleming
• [Asterisk-Users] zaptel and zapata configuration   Kevin P. Fleming
• [Asterisk-Users] attended transfer issue   Don Pobanz
• [Asterisk-Users] What's the best way to combine multiple VOIP lines into a single number?   Leo Burd
• [Asterisk-Users] Re: MeetMe: lots of buffer overruns/underruns when connecting over IAX   Tony Mountifield
• [Asterisk-Users] Asterisk FAX   Wasif
• [Asterisk-Users] Asterisk FAX   Aaron Daniel
• [Asterisk-Users] TDM2400P   blackgecko
• [Asterisk-Users] still some moh troubles   Matt Roth
• [Asterisk-Users] Announcement System for a Charity   Nabeel Jafferali
• [Asterisk-Users] enablling Te110p with PRI   Rafael Visser
• [Asterisk-Users] Suggestion Request: Coloc Provider in Chicago, IL area   S McGowan
• [Asterisk-Users] Announcement System for a Charity   Steve Jones
• [Asterisk-Users] Asterisk (RFC 3389)   Wasif
• [Asterisk-Users] still some moh troubles   Steve Totaro
• [Asterisk-Users] Announcement System for a Charity   Nabeel Jafferali
• [Asterisk-Users] TDM2400P   John Novack
• [Asterisk-Users] Announcement System for a Charity   Douglas Garstang
• [Asterisk-Users] attended transfer issue   John Novack
• [Asterisk-Users] Announcement System for a Charity   Nabeel Jafferali
• [Asterisk-Users] enablling Te110p with PRI   Steven Ringwald
• [Asterisk-Users] Re: MeetMe: lots of buffer overruns/underruns when connecting over IAX   Erik Hensema
• [Asterisk-Users] Announcement System for a Charity   Kerry Garrison
• [Asterisk-Users] Asterisk Won't start after SVN Trunk Update   Fred Noris
• [Asterisk-Users] Asterisk FAx-to-Email   Wasif
• [Asterisk-Users] Asterisk Won't start after SVN Trunk Update   Joshua Colp
• [Asterisk-Users] Asterisk Won't start after SVN Trunk Update   Brian Capouch
• [Asterisk-Users] Asterisk Won't start after SVN Trunk Update   Joshua Colp
• [Asterisk-Users] channels change names   Jon-o Addleman
• [Asterisk-Users] enablling Te110p with PRI   Rafael Visser
• [Asterisk-Users] Problem with TE110P configuration   Maximo Villamayor
• [Asterisk-Users] queues and the '*' key   Sean Kennedy
• [Asterisk-Users] Agents and Realtime   Carlos Chavez
• [Asterisk-Users] Asterisk Won't start after SVN Trunk Update   Fred Noris
• [Asterisk-Users] Background() and Read()   Douglas Garstang
• [Asterisk-Users] enablling Te110p with PRI   Kevin Bockman
• [Asterisk-Users] Polycom MWI   Kerry Garrison
• [Asterisk-Users] Call recording   Wai Wu
• [Asterisk-Users] Outbound calls are failing   nrbwpi at gmail.com
• [Asterisk-Users] Polycom MWI   Bill Gibbs
• [Asterisk-Users] Call recording   Jon-o Addleman
• [Asterisk-Users] Digium cards, so disappointing !   Boris Bakchiev
• [Asterisk-Users] Background() and Read()   Peter Fern
• [Asterisk-Users] channels change names   Peter Fern
• [Asterisk-Users] queues and the '*' key   Peter Fern
• [Asterisk-Users] channels change names   Gary Reuter
• [Asterisk-Users] Announcement System for a Charity   Steve Jones
• [Asterisk-Users] Call recording   Wai Wu
• [Asterisk-Users] MeetAsterisk in Europe - register today!   Olle E Johansson
• [Asterisk-Users] channels change names   Peter Fern
• [Asterisk-Users] Asterisk FAX   Damon Estep
• [Asterisk-Users] Colour coding the dialplan -- NoOp and ANSI codes?   Leif Madsen
• [Asterisk-Users] Digium cards, so disappointing !   Anton Krall
• [Asterisk-Users] Polycom MWI   Kerry Garrison
• [Asterisk-Users] Polycom MWI   Avi Miller
• [Asterisk-Users] Problem with TE110P configuration   Paulo Scardine
• [Asterisk-Users] T1 to cross connect remote PBX and asterisk   C F
• [Asterisk-Users] Get sysdate + 5 minutes   Arjan Kroon
• [Asterisk-Users] asterisk 1.2.7.1 crashing my newly built system   Rob Terhaar
• [Asterisk-Users] Modem connection   Tomislav Parčina
• [Asterisk-Users] Get sysdate + 5 minutes   Peter Fern
• [Asterisk-Users] Modem connection   Steve Underwood
• [Asterisk-Users] Voice mail issuse when pressing 0   Peter Fern
• [Asterisk-Users] Announcement System for a Charity   Michiel van Baak
• [Asterisk-Users] USB VoIP phone with G729 support   Obelix
• [Asterisk-Users] queues and the '*' key   Dinesh Nair
• [Asterisk-Users] Get sysdate + 5 minutes   Mark Ackroyd
• [Asterisk-Users] problem with sphinx2   serge messa
• [Asterisk-Users] Call recording   Mark Ackroyd
• [Asterisk-Users] some EICON Diva 4BRI questions   Klaus Darilion
• [Asterisk-Users] meetme scenario   Viktor Tatianin
• [Asterisk-Users] Cubix Softphone + Asterisk 1.2.6   Tim Panton
• [Asterisk-Users] Jingle support - can we test the feature ?   Tim Panton
• [Asterisk-Users] some EICON Diva 4BRI questions   David Waugh
• [Asterisk-Users] Real-time Database Front-end   James Nunnerley
• [Asterisk-Users] Re: Cisco 7960 6.3 unlock/reset?   Shaun
• [Asterisk-Users] some EICON Diva 4BRI questions   Klaus Darilion
• [Asterisk-Users] Re: Cisco 7960 6.3 unlock/reset?   Shaun
• [Asterisk-Users] Re: MACRO_RESULT=ABORT   Shaun
• [Asterisk-Users] Polycom MWI   Andrew Kohlsmith
• [Asterisk-Users] Unicall MFRC2 Problems with BrT.   Jefferson Carvalho
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] AAH or Fedora an Asterisk by sources   asterisk at mazone.info
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] some EICON Diva 4BRI questions   David Waugh
• [Asterisk-Users] AAH or Fedora an Asterisk by sources   Steve Totaro
• [Asterisk-Users] Asterisk on Red Hat AS 4?   Mimmus
• [Asterisk-Users] How to select Ceptral's Voice in Asterisk's Swift application??   Pimjai Wesnarat
• [Asterisk-Users] Asterisk on Red Hat AS 4?   Steve Totaro
• [Asterisk-Users] How to select Ceptral's Voice in Asterisk's Swiftapplication??   Steve Totaro
• [Asterisk-Users] Cubix Softphone + Asterisk 1.2.6   Steve Totaro
• [Asterisk-Users] Redirecting to another service/server   broadbandvoice at comcast.net
• [Asterisk-Users] record_in / record_out configuration parameters   Florian Muellner
• [Asterisk-Users] some EICON Diva 4BRI questions   Klaus Darilion
• [Asterisk-Users] Airspan / Arelnet GW and Asterisk   Michel Luczak
• [Asterisk-Users] some EICON Diva 4BRI questions   Klaus Darilion
• [Asterisk-Users] Real-time Database Front-end   Rich Adamson
• [Asterisk-Users] some EICON Diva 4BRI questions   Klaus Darilion
• [Asterisk-Users] Polycom MWI   Bill Gibbs
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] some EICON Diva 4BRI questions   Avi Miller
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] How to select Ceptral's Voice in Asterisk's Swift application??   Shane Young
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] problem with TE205   Augustine Olaifa
• [Asterisk-Users] Re: Modem connection   Tomislav Parčina
• [Asterisk-Users] Flash Panel / Queue Slots   Thomas Broda
• [Asterisk-Users] Polycom MWI   Kerry Garrison
• [Asterisk-Users] Polycom MWI   Sean Cook
• [Asterisk-Users] Separating Asterisk SIP extensions from dialing each other.   Rick Smith
• [Asterisk-Users] Unicall MFRC2 Problems with BrT.   Moises Silva
• [Asterisk-Users] problem with TE205   Ondrej Valousek
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   james.texter at cox.net
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   Andrew Latham
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   Steve Kennedy
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   Adam Lewis
• [Asterisk-Users] problem with TE205   Infobox Peru
• [Asterisk-Users] MoH issue   Kevin Smith
• [Asterisk-Users] Parallel Dial: Busy detection - stop when any is busy?   Pimjai Wesnarat
• [Asterisk-Users] problem with TE205   Remco Barende
• [Asterisk-Users] problem with TE205   Rob Lith
• [Asterisk-Users] Definitive list of sounds   Steve Kennedy
• [Asterisk-Users] HANGUPCAUSE on SIP channels   Eric Futch
• [Asterisk-Users] roundrobin strategy in queues not working as described?   Jim Rice
• [Asterisk-Users] SIP domain in Asterisk   Joao Pereira
• [Asterisk-Users] roundrobin strategy in queues not working as described?   Josué Conti
• [Asterisk-Users] Asterisk FAx-to-Email   Wasif
• [Asterisk-Users] problem with TE205   Remco Barende
• [Asterisk-Users] How to select Ceptral's Voice in Asterisk'sSwift application??   kevin ling
• [Asterisk-Users] Asterisk FAX-to-Email   Wasif
• [Asterisk-Users] Polycom MWI   Eric "ManxPower" Wieling
• [Asterisk-Users] channels change names   Jonathan Addleman
• [Asterisk-Users] Call recording   Jonathan Addleman
• [Asterisk-Users] some EICON Diva 4BRI questions   Olivier Krief
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   Chad Osmond
• [Asterisk-Users] MWI in multi-PBX setup   Olivier Krief
• [Asterisk-Users] wellgate FXO unit   Jerry Geis
• [Asterisk-Users] MWI in multi-PBX setup   C F
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   Geoff Manning
• [Asterisk-Users] Easier install of QueueMetrics on Asterisk@Home   Lenz
• [Asterisk-Users] some EICON Diva 4BRI questions   Jens Vagelpohl
• [Asterisk-Users] Separating Asterisk SIP extensions from dialing each other.   Jeremy Parr
• [Asterisk-Users] Jingle support - can we test the feature ?   Robert Rozman
• [Asterisk-Users] Parallel Dial: Busy detection - stop when any is busy?   Moises Silva
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   Andrew Latham
• [Asterisk-Users] Grandstream Budge Tone 101 keeps deregistering   Marcel Hecko
• [Asterisk-Users] Codec problem from SIP to H323   Alejandro Mejía Evertsz
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   Matt Roth
• [Asterisk-Users] Grandstream Budge Tone 101 keeps deregistering   Steve Jones
• [Asterisk-Users] still some moh troubles   Anthony Rodgers
• [Asterisk-Users] Asterisk on Red Hat AS 4?   Anthony Rodgers
• [Asterisk-Users] extension match sip address   Jon-o Addleman
• [Asterisk-Users] confused about iax and voip providers termination   T. Shaw
• [Asterisk-Users] Definitive list of sounds   Kristian Kielhofner
• [Asterisk-Users] Definitive list of sounds   Eric "ManxPower" Wieling
• [Asterisk-Users] extension match sip address   Eric "ManxPower" Wieling
• [Asterisk-Users] extension match sip address   Alejandro Mejía Evertsz
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   Kristian Kielhofner
• [Asterisk-Users] extension match sip address   Alejandro Mejía Evertsz
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   Cory Andrews
• [Asterisk-Users] test - ignore   Rich Adamson
• [Asterisk-Users] server choice   issam
• [Asterisk-Users] extension match sip address   Jon-o Addleman
• [Asterisk-Users] Very high size-32 usage   Anthony Rodgers
• [Asterisk-Users] Error installing oh323   Alejandro Mejía Evertsz
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   Mike Garey
• [Asterisk-Users] confused about iax and voip providers termination   Moises Silva
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   William M Conlon
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   Kevin P. Fleming
• [Asterisk-Users] Error installing oh323   T. Shaw
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   William M Conlon
• [Asterisk-Users] Unicall MFRC2 Problems with BrT.   Anton Krall
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   Steven Ringwald
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   Steven Ringwald
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   William M Conlon
• [Asterisk-Users] 1.2.7.1 on FC5 won't make install   Cliff Savage
• [Asterisk-Users] some EICON Diva 4BRI questions   Klaus Darilion
• [Asterisk-Users] Separating Asterisk SIP extensions from dialing each other.   Alex Mosburger
• [Asterisk-Users] Power over Ethernet (PoE) switch recommendations   stoffell
• [Asterisk-Users] Connecting to a cluster of SIP servers   Steve Hill
• [Asterisk-Users] 1.2.7.1 on FC5 won't make install   Alex Mosburger
• [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?   Louis-David Mitterrand
• [Asterisk-Users] RE: SPA 3000 - UK Replacement   Steven
• [Asterisk-Users] Connecting to a cluster of SIP servers   billy at kersting.com
• [Asterisk-Users] RE: SPA 3000 - UK Replacement   tom
• [Asterisk-Users] Sipura SP3000 question   RumaTech
• [Asterisk-Users] RE: SPA 3000 - UK Replacement   billy at kersting.com
• [Asterisk-Users] confused about iax and voip providers termination   Andrew Kohlsmith
• [Asterisk-Users] anybody get experience with dell powerconnect 3424 and QOS for asterisk traffic?   Jean-Louis curty
• [Asterisk-Users] Unicall MFRC2 Problems with BrT.   Moises Silva
• [Asterisk-Users] What about NCS and Asterisk?   Carlos Alberto Bernat Orozco
• [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?   Paul Mahler
• [Asterisk-Users] Re: what cable to connect a legacy PBX to a TE410P ?   Louis-David Mitterrand
• [Asterisk-Users] anybody get experience with dell powerconnect 3424 and QOS for asterisk traffic?   Brian Roy
• [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?   Steve Totaro
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Alexander Lopez
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   John Novack
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?"   Steve Totaro
• [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3   broadbandvoice at comcast.net
• [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone IncompatibleWith Asterisk 1.2.3   Steve Totaro
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?"   Alexander Lopez
• [Asterisk-Users] server choice   Steve Totaro
• [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone IncompatibleWith Asterisk 1.2.3   broadbandvoice at comcast.net
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?"   Paul Mahler
• [Asterisk-Users] Orative   Dovid Bender
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Andrew
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?"   Lacy Moore - Aspendora
• [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3   John Novack
• [Asterisk-Users] Sipura SP3000 question   Andres
• [Asterisk-Users] Sipura SP3000 question   Roshan Sembacuttiaratchy
• [Asterisk-Users] Sipura SP3000 question   Gonzalo Servat
• [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3   broadbandvoice at comcast.net
• [Asterisk-Users] Sipura SP3000 question   Rich Adamson
• [Asterisk-Users] How can I get a recording from a CD to my asterisk digital assistant   Davi-Ann
• [Asterisk-Users] Asterisk on FreeBSD + Passive ISDN BRI   Cian Hughes
• [Asterisk-Users] How can I get a recording from a CD to my asterisk digital assistant   Alberto Sagredo
• [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3   John Novack
• [Asterisk-Users] How can I get a recording from a CD to myasterisk digital assistant   Davi-Ann
• [Asterisk-Users] RE: SPA 3000 - UK Replacement   Wayne
• [Asterisk-Users] Sipura SP3000 question   Wayne
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ?"   Steven Totaro
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?"   Steven Totaro
• [Asterisk-Users] Don't see my post   broadbandvoice at comcast.net
• [Asterisk-Users] How to restrict simultaneous phone registrations   broadbandvoice at comcast.net
• [Asterisk-Users] No DTMF   broadbandvoice at comcast.net
• [Asterisk-Users] Don't see my post   Steven Totaro
• [Asterisk-Users] Need help with getting EXTEN from pstn hunt group   Jim Freeze
• [Asterisk-Users] Need help with getting EXTEN from pstn hunt group   Leo Ann Boon
• [Asterisk-Users] Unicall MFRC2 Problems with BrT.   Anton Krall
• [Asterisk-Users] TE410P card connection (was: Pinouts for T1/E1 crossover cable)   Louis-David Mitterrand
• [Asterisk-Users] 1.2.7.1 on FC5 won't make install   Cliff Savage
• [Asterisk-Users] Connecting to a cluster of SIP servers   Steve Hill
• [Asterisk-Users] Setting up a t38 fax gateway   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Codec G729 / x86_64 bits.   Jefferson Carvalho
• [Asterisk-Users] FritzCard, mISDN & "Anlagenanschluss"   Ralf Mueller
• [Asterisk-Users] Codec G729 / x86_64 bits.   Anderson Ling
• [Asterisk-Users] SIPredirect   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Accessing functions from AGI   Steve Hill
• [Asterisk-Users] Routing through ENUM   Steve Hill
• [Asterisk-Users] some EICON Diva 4BRI questions   Olivier Krief
• [Asterisk-Users] Accessing functions from AGI   Steve Totaro
• [Asterisk-Users] Need help with getting EXTEN from pstn hunt group   Jim Freeze
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] TE410P card connection (was: Pinouts for T1/E1crossover cable)   f6hqz-m at hamwlan.net
• [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?   Salah Eddine ELMRABET
• [Asterisk-Users] Accessing functions from AGI   Steve Hill
• [Asterisk-Users] Need help with getting EXTEN from pstn hunt group   Jim Freeze
• [Asterisk-Users] TE410P card connection (was: Pinouts for T1/E1crossover cable)   Steve Totaro
• [Asterisk-Users] Asterisk and SER hangup issue   Jon Farmer
• [Asterisk-Users] Need help with getting EXTEN from pstn hunt group   C F
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 21, Issue 130   Jordan Novak
• [Asterisk-Users] Need help with getting EXTEN from pstn hunt group   Alexander Lopez
• [Asterisk-Users] Codec G729 / x86_64 bits.   Hermann Wecke
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 21, Issue 130   Alexander Lopez
• [Asterisk-Users] E1 connexion   Steve Totaro
• [Asterisk-Users] Re: TDM2400P   blackgecko
• [Asterisk-Users] E1 connexion   issam
• [Asterisk-Users] asterisk at home, broadvoice and iptables   lenny at edpausa.com
• [Asterisk-Users] E1 connexion   Krzysztof Drewicz
• [Asterisk-Users] E1 connexion   Steve Totaro
• [Asterisk-Users] call queue problems   Dumpolid Exeplish
• [Asterisk-Users] New backport of T.38 fax passthrough functionality to asterisk-1.2.7.1   Adolfo R. Brandes
• [Asterisk-Users] Asterisk hangs up on incoming PSTN line to analog extension   Robert La Ferla
• [Asterisk-Users] Need help with getting EXTEN from pstn hunt group   Avi Miller
• [Asterisk-Users] Codec G729 / x86_64 bits.   Jefferson Carvalho
• [Asterisk-Users] Need help with getting EXTEN from pstn hunt group   Leo Ann Boon
• [Asterisk-Users] TE410P card connection   Leo Ann Boon
• [Asterisk-Users] Zap - Cahnnel bank - one way audio   Chris Mason (Lists)
• [Asterisk-Users] Zap - Cahnnel bank - one way audio   Alexander Lopez
• [Asterisk-Users] call queue problems   Kevin Smith
• [Asterisk-Users] Asteriak not starting with Ground Start Lines   Davi-Ann
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 21, Issue 132   jemmy_12345 frank
• [Asterisk-Users] SPA3000 in Singapore   KokMeng Loh
• [Asterisk-Users] Asterisk hangs up on incoming PSTN line to analog extension   Leonardo Silva
• [Asterisk-Users] Clearpath?   Michael Graves
• [Asterisk-Users] Flash Panel / Queue Slots   Nicolás Gudiño
• [Asterisk-Users] Asteriak not starting with Ground Start Lines   Eric "ManxPower" Wieling
• [Asterisk-Users] Re: TE410P card connection   Louis-David Mitterrand
• [Asterisk-Users] FritzCard, mISDN & "Anlagenanschluss"   Johann Steinwendtner
• [Asterisk-Users] 1/3 packets are reported dropped by tethereal   Abhimanyu Rapria
• [Asterisk-Users] sending special infoa fter login   René Enskat [Teamware GmbH]
• [Asterisk-Users] some EICON Diva 4BRI questions   Olivier Krief
• [Asterisk-Users] some EICON Diva 4BRI questions   Avi Miller
• [Asterisk-Users] annoying noise on analog phones on tdm400p   Thomas Artner
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] X100P Polarity Reverse Detection   Enky
• [Asterisk-Users] Playback(something,noanswer) on Zap?   Dmitry Ivanov
• [Asterisk-Users] MeetMe Call Out to invite   welemon lee
• [Asterisk-Users] some EICON Diva 4BRI questions   Klaus Darilion
• [Asterisk-Users] MeetMe Call Out to invite   Ben Q
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *   Olle E Johansson
• [Asterisk-Users] Re: MeetMe Call Out to invite   Tony Mountifield
• [Asterisk-Users] MeetMe Call Out to invite   Pimjai Wesnarat
• [Asterisk-Users] RE: SPA 3000 - UK Replacement   bails
• [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Dreadful results from zttest with TE210P and Dell 2850?   Remco Barende
• [Asterisk-Users] compiling zaptel-1.2.5   hgaillac-sip at yahoo.fr
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ?"   Mark Phillips
• [Asterisk-Users] Asterisk @ Home install on server with Apache already running   James Nunnerley
• [Asterisk-Users] Hi...Please help me   Crazy Boy
• [Asterisk-Users] Error messages   Tomislav Parčina
• [Asterisk-Users] 1.2.4/7 and chan_modem   Marnus van Niekerk
• [Asterisk-Users] Announcement System for a Charity   bails
• [Asterisk-Users] outbound calls to sip urls   Ajit
• [Asterisk-Users] 1.2.4/7 and chan_modem   tom
• [Asterisk-Users] Asterisk @ Home install on server with Apache already running   John covici
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Rich Adamson
• [Asterisk-Users] Quintum D3000   Neil Bullock
• [Asterisk-Users] compiling zaptel-1.2.5 [SOLVED]   hgaillac-sip at yahoo.fr
• [Asterisk-Users] SIPredirect [2]   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Setting up a t38 fax gateway [2]   hgaillac-sip at yahoo.fr
• [Asterisk-Users] User Defined VoiceMail announcement?   Benoit Panizzon
• [Asterisk-Users] Error messages   yusuf
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Leo Ann Boon
• [Asterisk-Users] answer delay   FaberK
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Rich Adamson
• Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Alexander Lopez
• [Asterisk-Users] Digium cards for sale   Senad Jordanovic
• [Asterisk-Users] Queue reload   Johann
• [Asterisk-Users] strange problem with Telasip DID, please help   Xin Li
• [Asterisk-Users] SPA-3000 Bug? Dropped calls while registering.   Dana Harding
• [Asterisk-Users] Announcement System for a Charity   Douglas Garstang
• [Asterisk-Users] Unicall MFRC2 Problems with BrT.   Moises Silva
• [Asterisk-Users] some EICON Diva 4BRI questions   Olivier Krief
• Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Colin Anderson
• [Asterisk-Users] outbound calls to sip urls   Jon-o Addleman
• [Asterisk-Users] User Defined VoiceMail announcement?   Eric "ManxPower" Wieling
• [Asterisk-Users] Hi...Please help me   Marcel Hecko
• [Asterisk-Users] Hi...Please help me   Marcel Hecko
• [Asterisk-Users] Announcement System for a Charity   Christopher Mayfield
• [Asterisk-Users] Help!!!!! DTMF detection is not working on Zap lines   Wai Wu
• [Asterisk-Users] Asteriak not starting with Ground Start Lines   John Novack
• [Asterisk-Users] Queue reload   BJ Weschke
• Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Rich Adamson
• [Asterisk-Users] User Defined VoiceMail announcement?   C F
• Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Dave Weis
• [Asterisk-Users] X100P support on POTS around the world (Slovakia)   Marcel Hecko
• [Asterisk-Users] Connecting to a cluster of SIP servers   Douglas Garstang
• [Asterisk-Users] fax and URA   Dov Bigio
• Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossovercable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to aTE410P ?"   Alexander Lopez
• [Asterisk-Users] Dreadful results from zttest with TE210P and Dell2850?   Craig Guy
• [Asterisk-Users] Faster Sound Files   Douglas Garstang
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover   Ken Godee
• [Asterisk-Users] Connecting to a cluster of SIP servers   Sergio García Murillo
• [Asterisk-Users] fxotune Problem   roly at redcetus.com
• [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)   Thomas Winter
• [Asterisk-Users] Dreadful results from zttest with TE210P and Dell2850?   Remco Barende
• [Asterisk-Users] Connecting to a cluster of SIP servers   Douglas Garstang
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover   Andrew Kohlsmith
• [Asterisk-Users] getting listed in Directory Assistance, the phone book   Bill Gibbs
• [Asterisk-Users] Connecting to a cluster of SIP servers   Aaron Daniel
• [Asterisk-Users] SMP kernel on Pent 4?   Rich Adamson
• [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *   Kevin P. Fleming
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Kevin P. Fleming
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover   Kevin P. Fleming
• [Asterisk-Users] SMP kernel on Pent 4?   Aaron Daniel
• [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)   Doug Lytle
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover   Alexander Lopez
• [Asterisk-Users] Connecting to a cluster of SIP servers   Douglas Garstang
• [Asterisk-Users] Connecting to a cluster of SIP servers   Andrew Kohlsmith
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover   Rich Adamson
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Andrew Kohlsmith
• [Asterisk-Users] Faster Sound Files   Jon-o Addleman
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?"   Alexander Lopez
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover   Alexander Lopez
• [Asterisk-Users] Question about Asterisk realtime   Tielin Xu
• [Asterisk-Users] CallerID/variable setting.   Ken D'Ambrosio
• [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Connecting to a cluster of SIP servers   Aaron Daniel
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover   Steve Underwood
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover   Andrew Kohlsmith
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover   Andrew Kohlsmith
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover   Kevin P. Fleming
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover   Michael Collins
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover   Kevin P. Fleming
• [Asterisk-Users] Asterisk to Linphone sound playback delay, and then choppy   Adam Ward
• [Asterisk-Users] Zap channels not disconnecting after PSTN line hangs up (getting empty voicemails)   Mike Garey
• [Asterisk-Users] Faster Sound Files   Douglas Garstang
• [Asterisk-Users] eyeBeam 1.5   Adrian A
• [Asterisk-Users] X100P support on POTS around the world (Slovakia)   Krzysztof Drewicz
• [Asterisk-Users] TDM2400P   Mojo with Horan & Company, LLC
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Rich Adamson
• [Asterisk-Users] Time out if channel does not ring   Cavanna, Richard
• [Asterisk-Users] HINTS with Polycom stops working after asterisk reload   Bartosz Jozwiak
• [Asterisk-Users] call queue problems   Dumpolid Exeplish
• [Asterisk-Users] X100P Polarity Reverse Detection   John Novack
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover   Andrew Kohlsmith
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover   Bart Fisher
• [Asterisk-Users] CallerID/variable setting.   Anthony Rodgers
• [Asterisk-Users] X100P support on POTS around the world (Slovakia)   Marcel Hecko
• [Asterisk-Users] HINTS with Polycom stops working after asterisk reload   sathvik sricharan
• [Asterisk-Users] HINTS with Polycom stops working after aster isk reload   Colin Anderson
• [Asterisk-Users] CallerID/variable setting.   Brent Torrenga
• [Asterisk-Users] wellgate FXO unit   Artifex Maximus
• [Asterisk-Users] Some questions re. T1 cards & QoS   hugolivude
• [Asterisk-Users] Channel Restart and Dropped calls   Chris Gamble
• [Asterisk-Users] SUSE 9.3, modprobe, and zaptel   Joe
• [Asterisk-Users] Some questions re. T1 cards & QoS   Cory Andrews
• [Asterisk-Users] HINTS with Polycom stops working after asterisk reload   Douglas Garstang
• [Asterisk-Users] Some questions re. T1 cards & QoS   Kevin P. Fleming
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover   Rich Adamson
• [Asterisk-Users] HINTS with Polycom stops working after asterisk reload   Kevin P. Fleming
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover   Andrew Kohlsmith
• [Asterisk-Users] Question about Asterisk realtime   Avi Miller
• [Asterisk-Users] Zap channels not disconnecting after PSTN line hangs up (getting empty voicemails)   Hadley Rich
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover   Rich Adamson
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover   Don Pobanz
• [Asterisk-Users] E1 testing   Andrew Nowrot
• [Asterisk-Users] this is just a post test   Tim Ferguson
• [Asterisk-Users] Hold button   Carlos Alperin
• [Asterisk-Users] Quintum D3000   Steve Totaro
• [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)   Mike Garey
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: PinoutsforT1/E1 crossover   Michael Collins
• [Asterisk-Users] Hold button   Jerry Jones
• [Asterisk-Users] Re: Shielding of T1/E1 cables WAS RE: PinoutsforT1/E1 crossover   Rich Adamson
• [Asterisk-Users] Question about Asterisk realtime   Tielin Xu
• [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)   Rich Adamson
• [Asterisk-Users] Two asterisk process in one hardware.   Juan Salas
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Sangoma Techdesk
• [Asterisk-Users] chan_gsm_bt Impression   Il Neofita
• [Asterisk-Users] answer delay   Peter J Dean
• [Asterisk-Users] Question about Asterisk realtime   Avi Miller
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Thomas Winter
• [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)   Thomas Winter
• [Asterisk-Users] answer delay   FaberK
• [Asterisk-Users] answer delay   FaberK
• [Asterisk-Users] Playback(something,noanswer) on Zap?   FaberK
• [Asterisk-Users] answer delay   Peter J Dean
• [Asterisk-Users] answer delay   Kevin P. Fleming
• [Asterisk-Users] answer delay   FaberK
• [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)   Mark Phillips
• [Asterisk-Users] Change name User-Agent   Will Velez
• [Asterisk-Users] Connecting to a cluster of SIP servers   Steve Hill
• [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)   Mike Garey
• [Asterisk-Users] Change name User-Agent   Joshua Colp
• [Asterisk-Users] [Issue] Does the *-pbx cmd page honour the absolute timeout value?   Peter J Dean
• [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)   John Novack
• [Asterisk-Users] [Issue] Does the *-pbx cmd page honour the absolute timeout value?   Kevin P. Fleming
• [Asterisk-Users] Two asterisk process in one hardware.   Josué Conti
• [Asterisk-Users] codec variable for incoming calls   J Shaun Hofer
• [Asterisk-Users] Faster Sound Files   Michael Wallette
• [Asterisk-Users] Two asterisk process in one hardware.   Andrew D Kirch
• [Asterisk-Users] Two asterisk process in one hardware.   Mike Fedyk
• [Asterisk-Users] SMP kernel on Pent 4?   Mike Fedyk
• [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)   Sean Cook
• [Asterisk-Users] Channel Restart and Dropped calls   issam
• [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)   Mike Garey
• [Asterisk-Users] Unicall MFRC2 Problems with BrT.   Anton Krall
• [Asterisk-Users] Re: Some questions re. T1 cards & QoS   hugolivude
• [Asterisk-Users] Asterisk2Billing   Scheda
• [Asterisk-Users] (no subject)   rommel malana
• [Asterisk-Users] Asterisk2Billing   billy at kersting.com
• [Asterisk-Users] (no subject)   C F
• [Asterisk-Users] Dialing Ring Groups from the Digital Receptionist-   Maxx Lobo
• [Asterisk-Users] Gsm Gateway , again !   Mohammad Salaque
• [Asterisk-Users] call queue problems   Kevin Smith
• [Asterisk-Users] Polycom Delay   Kevin Smith
• [Asterisk-Users] A@H 2.6 : problem connecting call from PSTN   WALTER LOH
• [Asterisk-Users] Polycom Delay   Gabriel Afana
• [Asterisk-Users] Development news :: New AEL and configuration system   Olle E Johansson
• [Asterisk-Users] Two asterisk process in one hardware.   Dmitry Ivanov
• [Asterisk-Users] Asterisk2Billing   Vahan Yerkanian
• [Asterisk-Users] Two asterisk process in one hardware.   Armin Schindler
• [Asterisk-Users] CHANUNAVAIL, busy and congestion   Joseph Rothstein
• [Asterisk-Users] billing realtime   random cluster
• [Asterisk-Users] Dialing Ring Groups from the Digital Receptionist-   kevin ling
• [Asterisk-Users] wellgate FXO unit   kevin ling
• [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)   Olle E Johansson
• [Asterisk-Users] E1 testing   Andrew Nowrot
• [Asterisk-Users] PRI got event: HDLC Bad FCS (8) on Primary D-channel of span   Nico Giefing
• [Asterisk-Users] Background asynchronous AGI   Tony Mountifield
• [Asterisk-Users] annoying noise on analog phones on tdm400p   Thomas Artner
• [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)   Thomas Winter
• [Asterisk-Users] Background asynchronous AGI   random cluster
• [Asterisk-Users] Another undefined pri_restart failure   Fred Noris
• [Asterisk-Users] No sound in one calling direction, men using PRI with E1 and Q.SIG   Peter Olsson
• [Asterisk-Users] Festival , Cannot hear the words after ","   John Joseph
• [Asterisk-Users] SQL update failing/long fullcontact   Mark Drayton
• [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)   Rich Adamson
• [Asterisk-Users] MeetAsterisk in Europe - register today!   fbergeret at ges.fr
• [Asterisk-Users] SMP kernel on Pent 4?   Rich Adamson
• [Asterisk-Users] res_perl voor asterisk 1.2.4   Arjan Kroon
• [Asterisk-Users] Voicemail being cut-off   Andre Courchesne - Consultant
• [Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly (UPDATE)   Dave Fullerton
• [Asterisk-Users] SIP HEADER FROM: without CALLERID(name)   John Novack
• [Asterisk-Users] Lastest stable build   Wai Wu
• [Asterisk-Users] Lastest stable build   Olle E Johansson
• [Asterisk-Users] Really Old Rotary Phone   Sean Cook
• [Asterisk-Users] Really Old Rotary Phone   Jerry Jones
• [Asterisk-Users] Really Old Rotary Phone   John Novack
• [Asterisk-Users] Two asterisk process in one hardware.   Juan Salas
• [Asterisk-Users] Re: Some questions re. T1 cards & QoS   Kevin P. Fleming
• [Asterisk-Users] About Softphone IAX free for Pocket PC   makevuy
• [Asterisk-Users] Really Old Rotary Phone   Sean Cook
• [Asterisk-Users] Really Old Rotary Phone   Rusty Dekema
• [Asterisk-Users] About Softphone IAX free for Pocket PC   Kerry Garrison
• [Asterisk-Users] Really Old Rotary Phone   Sean Cook
• [Asterisk-Users] Updated: No audio when dialing in via PRI with Q.SIG   Peter Olsson
• [Asterisk-Users] Really Old Rotary Phone   John Novack
• [Asterisk-Users] Question on connecting to another system   Matt
• [Asterisk-Users] Really Old Rotary Phone   John Novack
• [Asterisk-Users] About Softphone IAX free for Pocket PC   Rusty Dekema
• [Asterisk-Users] Updated: No audio when dialing in via PRI withQ.SIG   Alexander Lopez
• [Asterisk-Users] About Softphone IAX free for Pocket PC   Alexander Lopez
• [Asterisk-Users] About Softphone IAX free for Pocket PC   Robert Augustyn
• Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Andrew Latham
• [Asterisk-Users] About Softphone IAX free for Pocket PC   Steve Underwood
• [Asterisk-Users] TDM400P: flash on analog phones doesn't work   Patrick
• [Asterisk-Users] SMS to call back   Jeremy
• [Asterisk-Users] Background asynchronous AGI   Matt
• [Asterisk-Users] Unicall MFC problems in 0.0.3+asterisk 1.2   Guillermo Freige
• [Asterisk-Users] MFCR2 in Brazil, someone?   Moises Silva
• [Asterisk-Users] About Softphone IAX free for Pocket PC   Robert Augustyn
• SV: [Asterisk-Users] Updated: No audio when dialing in via PRIwithQ.SIG   Peter Olsson
• [Asterisk-Users] Auto Logout from queue   Kerry Garrison
• [Asterisk-Users] Sip t38 gateway tests   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Really Old Rotary Phone   Sean Cook
• [Asterisk-Users] Auto Logout from queue   Alexander Lopez
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 21, Issue 132   jemmy_12345 frank
• [Asterisk-Users] Channel Restart and Dropped calls   Chris Gamble
• [Asterisk-Users] Help on chan_misdn and MSN's   Cosmin Prund
• [Asterisk-Users] Really Old Rotary Phone   John Novack
• [Asterisk-Users] FastAGI Connection Failure and Hangup   Steve Totaro
• [Asterisk-Users] Help with using Asterisk with PlusNet in the UK   Monty
• [Asterisk-Users] Channel Restart and Dropped calls   stoffell
• [Asterisk-Users] Splitting Zap channels into trunks?   Kerry Garrison
• [Asterisk-Users] One Way Audio....in the middle of a call   Geoff Manning
• [Asterisk-Users] Auto Logout from queue   Kerry Garrison
• [Asterisk-Users] One Way Audio....in the middle of a call   Frederic Jean
• [Asterisk-Users] TE410 and 411   Carlos Chavez
• [Asterisk-Users] TE410 and 411   Kevin P. Fleming
• [Asterisk-Users] queues that do not play music   Andre Ruiz
• [Asterisk-Users] Auto Logout from queue   Alberto Sagredo
• [Asterisk-Users] Sphinx   Douglas Garstang
• [Asterisk-Users] Auto Logout from queue   Christopher Mayfield
• [Asterisk-Users] Touch tone recognition issues   Bryan Mahin
• [Asterisk-Users] Sip t38 gateway tests   C F
• [Asterisk-Users] Pressing ## end the call and return to menu   shirali
• [Asterisk-Users] 56K Dialup and VOIP over same PRIs   Ian White
• [Asterisk-Users] One Way Audio....in the middle of a call   Philip Edelbrock
• [Asterisk-Users] 56K Dialup and VOIP over same PRIs   George Pajari
• [Asterisk-Users] Music on Hold bug? User disconnect Sip user agent   Marco Mouta
• [Asterisk-Users] 56K Dialup and VOIP over same PRIs   Damon Estep
• [Asterisk-Users] 56K Dialup and VOIP over same PRIs   Kevin P. Fleming
• [Asterisk-Users] One Way Audio....in the middle of a call   Dan Levy
• [Asterisk-Users] Agents <--> Extensions   Shaun
• [Asterisk-Users] Re: FastAGI Connection Failure and Hangup   Matt King
• [Asterisk-Users] Touch tone recognition issues   John Novack
• [Asterisk-Users] USB conference phone   Dean Collins
• [Asterisk-Users] Here I am facing problem of Voice Breakage   Mr shobhit nirala
• [Asterisk-Users] Hi...Please help me   Paul Hales
• [Asterisk-Users] Trying to set up automatic announcement upon transfer for IVR in AAH 2.8   Carl Youngblood
• [Asterisk-Users] Trying to set up automatic announcement upon   Mike Gach
• [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)   Eric "ManxPower" Wieling
• [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)   Eric "ManxPower" Wieling
• [Asterisk-Users] test numbers in different countries!   Ronald Wiplinger
• [Asterisk-Users] MFCR2 in Brazil, someone?   Melcon Moraes
• [Asterisk-Users] test numbers in different countries!   Jason Frisch
• [Asterisk-Users] Hi...Please help me   Gonzalo Servat
• [Asterisk-Users] billing realtime   JP Carballo
• [Asterisk-Users] Background asynchronous AGI   JP Carballo
• [Asterisk-Users] billing realtime   Jon Farmer
• [Asterisk-Users] TDM400P: flash on analog phones doesn't work   Leo Ann Boon
• [Asterisk-Users] billing realtime   Darren Wiebe
• [Asterisk-Users] Need some help on queues with agents(SIP members) with multiple phones.   Arne Morten Johansen
• [Asterisk-Users] billing realtime   Nick Hoffman
• [Asterisk-Users] AGI and incoming call   Olivier Saulnier
• SV: [Asterisk-Users] Need some help on queues with agents(SIP members)with multiple phones.   Arne Morten Johansen
• [Asterisk-Users] billing realtime   JP Carballo
• [Asterisk-Users] billing realtime   Senad Jordanovic
• [Asterisk-Users] CISCO 7960G - SIP Configuration   sdcharly at gmail.com
• [Asterisk-Users] # and call speed   Pasqualotto Enrico
• [Asterisk-Users] Intergrate Asterisk IP PBX with Legacy PBX, continuing existing funtionality of legacy pbx   Vidura Senadeera
• [Asterisk-Users] Re: billing realtime   Tony Mountifield
• [Asterisk-Users] AGI and incoming call   Steve Totaro
• [Asterisk-Users] Re: billing realtime   Tony Mountifield
• [Asterisk-Users] AGI and incoming call   Olivier Saulnier
• [Asterisk-Users] billing realtime   random cluster
• [Asterisk-Users] Stange behaviour on 4port BRI Asterisk 1.0.10-BRIstuffed-0.2.0-RC8q   Alasdair Gow
• [Asterisk-Users] Re: billing realtime   Senad Jordanovic
• [Asterisk-Users] test numbers in different countries!   Dmitry Ivanov
• [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inbound call   Giorgio Incantalupo
• [Asterisk-Users] test numbers in different countries!   James Nunnerley
• [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inbound call   Hadley Rich
• [Asterisk-Users] Avoiding deadlock... Problem   manish
• [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inbound call   Giorgio Incantalupo
• [Asterisk-Users] TDM400P: flash on analog phones doesn't work   Patrick
• [Asterisk-Users] Sip Phones with BLA Support   Tim Ferguson
• [Asterisk-Users] Sip Phones with BLA Support   Steve Langstaff
• [Asterisk-Users] do extensions must be numbers in Asterisk@Home?   Joao Pereira
• [Asterisk-Users] Re: Agents <--> Extensions   Shaun
• [Asterisk-Users] queues that do not play music   Mike
• [Asterisk-Users] Hi...Please help me   Crazy Boy
• [Asterisk-Users] 7960G SIP Issue   sdcharly at gmail.com
• [Asterisk-Users] Re: Intergrate Asterisk IP PBX with Legacy PBX, continuing existing funtionality of legacy pbx   Steven
• [Asterisk-Users] 7960G SIP Issue   Joe Greco
• [Asterisk-Users] Re: # and call speed   Steven
• [Asterisk-Users] 7960G SIP Issue   Steve Blair
• [Asterisk-Users] Registering to H.323 Cisco gatekeeper   Nils-Anders Duesund Nøttseter
• [Asterisk-Users] Splitting Zap channels into trunks?   Rich Adamson
• [Asterisk-Users] astcc: need partial pin code   Ronald Wiplinger
• [Asterisk-Users] 2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my *   Andy Green
• [Asterisk-Users] Sphinx2   Ronald Wiplinger
• [Asterisk-Users] annoying noise on analog phones on tdm400p   Eric "ManxPower" Wieling
• [Asterisk-Users] Another undefined pri_restart failure   Eric "ManxPower" Wieling
• [Asterisk-Users] I am looking for a webphone on MY SITE   Ronald Wiplinger
• [Asterisk-Users] test numbers in different countries!   Dmitry Ivanov
• [Asterisk-Users] Touch tone recognition issues   Eric "ManxPower" Wieling
• [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inbound call   Eric "ManxPower" Wieling
• [Asterisk-Users] ODBC Storage for voicemail messages in database   Dan Littlejohn
• [Asterisk-Users] 2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my *   picciuX
• [Asterisk-Users] IAX calls dropping after minutes   Chris Mason (Lists)
• [Asterisk-Users] Configuring QoS Params in UIP-200   Waldo Rubinstein
• [Asterisk-Users] USB conference phone   Jim Houser
• [Asterisk-Users] billing realtime   Jon Farmer
• [Asterisk-Users] RE: SOLVED: No audio when dialing in via PRI with Q.SIG   Peter Olsson
• [Asterisk-Users] Re: billing realtime   Jon Farmer
• [Asterisk-Users] MFCR2 in Brazil, someone?   Moises Silva
• [Asterisk-Users] IAX calls dropping after minutes   Rich Adamson
• [Asterisk-Users] MFCR2 in Brazil, someone?   Moises Silva
• Pinouts for T1/E1 crossover cable WAS "RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?"   Dinesh Nair
• [Asterisk-Users] No Caller-ID With Cisco PAP2T-NA   Matt
• [Asterisk-Users] I am looking for a webphone on MY SITE   Jim Houser
• RES: [Asterisk-Users] MFCR2 in Brazil, someone?   Virmones Pereira Tavares de Miranda
• [Asterisk-Users] Re: [Serusers] Sip t38 gateway tests   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Hi...Please help me   Evalyn Wafula
• [Asterisk-Users] Re: billing realtime   Tony Mountifield
• [Asterisk-Users] MFCR2 in Brazil, someone?   Steve Underwood
• [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)   Mike Garey
• [Asterisk-Users] Camp on?   Patrick
• [Asterisk-Users] ODBC Storage for voicemail messages in database   Noah Miller
• [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)   John Novack
• [Asterisk-Users] Stuck in Queues   tom
• [Asterisk-Users] USB conference phone   Kerry Garrison
• [Asterisk-Users] I am looking for a webphone on MY SITE   Tom Hayden
• [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inboundcall   kevin ling
• [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inboundcall   kevin ling
• [Asterisk-Users] Polycom Delay   Noah Miller
• [Asterisk-Users] No Caller-ID With Cisco PAP2T-NA   Tom Vile
• [Asterisk-Users] About Softphone IAX free for Pocket PC   Steve Jones
• [Asterisk-Users] Help on chan_misdn and MSN's   Cosmin Prund
• [Asterisk-Users] AGI and incoming call   Innocent Evil
• [Asterisk-Users] Stuck in Queues   Time Bandit
• [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inboundcall   Tom Vile
• [Asterisk-Users] I am looking for a webphone on MY SITE   Ronald Wiplinger
• [Asterisk-Users] kernel - module problem   Giuseppe Parlato
• [Asterisk-Users] I am looking for a webphone on MY SITE   Bruce Reeves
• [Asterisk-Users] I am looking for a webphone on MY SITE   Jim Houser
• [Asterisk-Users] Early media after a dial command   Benjamin Lawetz
• [Asterisk-Users] USB conference phone   Dean Collins
• [Asterisk-Users] I am looking for a webphone on MY SITE   Dean Collins
• [Asterisk-Users] Stuck in Queues   tom
• [Asterisk-Users] Camp on?   Andreas Sikkema
• [Asterisk-Users] No Caller-ID With Cisco PAP2T-NA   Matt
• [Asterisk-Users] AGI and incoming call   Olivier Saulnier
• [Asterisk-Users] 2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my *   Time Bandit
• [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)   Mike Garey
• [Asterisk-Users] I am looking for a webphone on MY SITE   Jim Houser
• [Asterisk-Users] Re: billing realtime   Jon Farmer
• [Asterisk-Users] Camp on?   Rob Hillis
• [Asterisk-Users] USB conference phone   Jim Houser
• [Asterisk-Users] Camp on?   Jon Farmer
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 21, Issue 132   jemmy_12345 frank
• [Asterisk-Users] Recording queue transfers   Maximiliano J. Goldsmid
• [Asterisk-Users] I am looking for a webphone on MY SITE   Ronald Wiplinger
• [Asterisk-Users] No Caller-ID With Cisco PAP2T-NA   Tom Vile
• [Asterisk-Users] (Wanted OEM VOIP and SKYPE PRODUCTS)   VOICEIN at aol.com
• [Asterisk-Users] asterisk no longer compiles on gcc 2.95   Steve Hanselman
• [Asterisk-Users] How to configure Asterisk to handle multiple Gizmo accounts?   Leo Burd
• [Asterisk-Users] Explain to me VoIP termination service.   Clint Willard
• [Asterisk-Users] I am looking for a webphone on MY SITE   Tim Panton
• [Asterisk-Users] USB conference phone   Dean Collins
• [Asterisk-Users] I am looking for a webphone on MY SITE   Infobox Peru
• [Asterisk-Users] Callback help   Wasif
• [Asterisk-Users] cannot transfer to call waiting call on ip500   PCSUPPORT PC SUPPORT
• [Asterisk-Users] asterisk no longer compiles on gcc 2.95   Jay Milk
• [Asterisk-Users] cannot transfer to call waiting call on ip500   Andrew Kohlsmith
• [Asterisk-Users] USB conference phone   Steve Feinstein
• [Asterisk-Users] I am looking for a webphone on MY SITE   Dean Collins
• [Asterisk-Users] Question about the zaptel-1.2.5-patch   Wai Wu
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   Benchev
• [Asterisk-Users] About Softphone IAX free for Pocket PC   Robert Augustyn
• [Asterisk-Users] I am looking for a webphone on MY SITE   Tim Panton
• [Asterisk-Users] billing realtime   Josh McAllister
• [Asterisk-Users] Question about the zaptel-1.2.5-patch   Thomas Artner
• [Asterisk-Users] Phone Emergency - Need IAX Help   David Tillman
• [Asterisk-Users] A@H and channel announcement   Francesco Peeters
• [Asterisk-Users] Help! * Won't Start after SVN Trunk Update - SuSE 10   Fred Noris
• [Asterisk-Users] clipcomm versus sipura/linksys   Casey Boone
• [Asterisk-Users] Voicemaster   Kresimir Petrovic
• [Asterisk-Users] I am looking for a webphone on MY SITE   Robert Rozman
• [Asterisk-Users] Re: billing realtime   Tony Mountifield
• [Asterisk-Users] some EICON Diva 4BRI questions   Klaus Darilion
• [Asterisk-Users] Cisco 7970 SIP - few questions   Omar A. Sabek
• [Asterisk-Users] Status of Queue   Kevin Savoy
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] Status of Queue   Alexander Lopez
• [Asterisk-Users] cell mobile network (GSM) to Asterisk   Marcel Hecko
• [Asterisk-Users] Status of Queue   BJ Weschke
• [Asterisk-Users] Hook Flash via SIP INFO command?   Bryan McLellan
• [Asterisk-Users] Sphinx2   Mojo with Horan & Company, LLC
• [Asterisk-Users] MFCR2 in Brazil, someone?   Melcon Moraes
• [Asterisk-Users] USB conference phone   Kerry Garrison
• [Asterisk-Users] Problems with Eicon Diva V-4BRI - 2nd Port   Avi Miller
• [Asterisk-Users] Pattern matching problem   hugolivude
• [Asterisk-Users] USB conference phone   Dean Collins
• [Asterisk-Users] USB conference phone   Kerry Garrison
• [Asterisk-Users] Pattern matching problem   Eric "ManxPower" Wieling
• [Asterisk-Users] USB conference phone   John covici
• [Asterisk-Users] Problems with Eicon Diva V-4BRI - 2nd Port   Avi Miller
• [Asterisk-Users] Re: Pattern matching problem   hugolivude
• [Asterisk-Users] Problem with a TDM-400P   Nigel Smith
• [Asterisk-Users] Re: Pattern matching problem   Eric "ManxPower" Wieling
• [Asterisk-Users] Re: Pattern matching problem   Waldo Rubinstein
• [Asterisk-Users] Re: Pattern matching problem   hugolivude
• [Asterisk-Users] Re: Pattern matching problem   Andrew Kohlsmith
• [Asterisk-Users] Camp on?   dthurn at tstt.net.tt
• [Asterisk-Users] Re: Pattern matching problem   kevin ling
• [Asterisk-Users] no audio for ring group.   Terry Denovan
• [Asterisk-Users] Need Help! --> Incoming calls are just ringing and not hitting the Digital Receptionist   dthurn at tstt.net.tt
• [Asterisk-Users] Re: Pattern matching problem   Eric "ManxPower" Wieling
• [Asterisk-Users] Re: Pattern matching problem   hugolivude
• [Asterisk-Users] Re: Pattern matching problem   hugolivude
• [Asterisk-Users] Re: Pattern matching problem   Andrew Kohlsmith
• [Asterisk-Users] Camp on?   Eric "ManxPower" Wieling
• [Asterisk-Users] Paging on Aastra analog phones.   Richard Schroeder
• [Asterisk-Users] Paging on Aastra analog phones.   Kerry Garrison
• [Asterisk-Users] Asterisk as a phone survey system   TV JOE
• [Asterisk-Users] Asterisk as a phone survey system   Kerry Garrison
• [Asterisk-Users] Re: Pattern matching problem   Aaron Daniel
• [Asterisk-Users] Camp on?   Nathan Alberti
• [Asterisk-Users] Background asynchronous AGI   Anton Krall
• [Asterisk-Users] Unable to accept incoming PSTN calls   Johnny Stork
• [Asterisk-Users] func_odbc and 1.2.7.1   Jon Webster
• [Asterisk-Users] getting asterisk to reliably answer a voip line   jnuoiqweahf kajhdsff
• [Asterisk-Users] treating an incoming call as a local extension   jnuoiqweahf kajhdsff
• [Asterisk-Users] Accessing PARKEDAT variable in AGI   jnuoiqweahf kajhdsff
• [Asterisk-Users] replacing step-by-step giving echo   stoffell
• [Asterisk-Users] call queue problems   Dumpolid Exeplish
• [Asterisk-Users] Extreme delay before * processes call files   Remco Barende
• [Asterisk-Users] treating an incoming call as a local extension   kevin ling
• [Asterisk-Users] Re: Pattern matching problem   kevin ling
• [Asterisk-Users] Re: Cisco 7970 SIP - few questions   Tomislav Parčina
• SV: [Asterisk-Users] treating an incoming call as a local extension   Arne Morten Johansen
• [Asterisk-Users] Re: Pattern matching problem   kevin ling
• [Asterisk-Users] SATA hard disk compatibility   amna saleem
• [Asterisk-Users] Camp on?   Andreas Sikkema
• [Asterisk-Users] SMP kernel on Pent 4?   Tomas Stribrny
• [Asterisk-Users] Re: Pattern matching problem   kevin ling
• [Asterisk-Users] astcc: need partial pin code   Ronald Wiplinger
• [Asterisk-Users] Asterisk Hangs the whole system   A.R. Nasir Qureshi
• [Asterisk-Users] AGI and incoming call   picciuX
• [Asterisk-Users] SATA hard disk compatibility   Assaf Flatto
• [Asterisk-Users] GXP-2000: disable provisioning   Mimmus
• [Asterisk-Users] getting asterisk to reliably answer a voip line   picciuX
• [Asterisk-Users] 1.2.4/7 and chan_modem   Marnus van Niekerk
• [Asterisk-Users] Asterisk Voice Problems   Shyam Gopale
• [Asterisk-Users] Asterisk Voice Problems   kevin ling
• [Asterisk-Users] Asterisk to Dial a number , after getting a mail notification ,   John Joseph
• [Asterisk-Users] Asterisk to Dial a number , after getting a mail notification ,   picciuX
• [Asterisk-Users] AGI and incoming call   Olivier Saulnier
• [Asterisk-Users] Camp on?   Patrick
• [Asterisk-Users] Camp on?   Patrick
• [Asterisk-Users] some EICON Diva 4BRI questions   Klaus Darilion
• [Asterisk-Users] Camp on?   Nathan Alberti
• [Asterisk-Users] treating an incoming call as a local extension   jnuoiqweahf kajhdsff
• [Asterisk-Users] astcc: need partial pin code   Benchev
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] Asterisk as a phone survey system   Steve Totaro
• [Asterisk-Users] Autodial feature doesn't return $DIALSTATUS values   elloallo at infinito.it
• [Asterisk-Users] Help on chan_misdn and MSN's   Amatisoft SRL
• [Asterisk-Users] (no subject)   Dovid Bender
• [Asterisk-Users] getting asterisk to reliably answer a voip line   jnuoiqweahf kajhdsff
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   Dovid Bender
• [Asterisk-Users] Anyone using the GSMgateway from CyberTelecom ?   Dovid Bender
• [Asterisk-Users] Asterisk as a phone survey system   Dovid Bender
• [Asterisk-Users] Asterisk Hangs the whole system   Dovid Bender
• [Asterisk-Users] zt_pri-error   Christian Gansberger
• [Asterisk-Users] Asterisk IVR / Scalability   Dovid Bender
• [Asterisk-Users] Accessing PARKEDAT variable in AGI   Andrew Kohlsmith
• [Asterisk-Users] Camp on?   Patrick
• [Asterisk-Users] billing realtime   Dovid Bender
• [Asterisk-Users] Billing Server Open Source   Dovid Bender
• [Asterisk-Users] Re: Pattern matching problem   Andrew Kohlsmith
• [Asterisk-Users] Camp on?   Rich Adamson
• [Asterisk-Users] astcc: need partial pin code   Benchev
• [Asterisk-Users] Unable to accept incoming PSTN calls   Dovid Bender
• [Asterisk-Users] SMP kernel on Pent 4?   Rich Adamson
• [Asterisk-Users] Interesting Dial-Plan Question   Matt
• [Asterisk-Users] Asterisk Hangs the whole system   Rich Adamson
• [Asterisk-Users] URGENTS: seek people for video tests with asterisk/ser/rtpproxy + eyebeam   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Need help configuring Asterisk with Alepo   info at dimal.com.al
• [Asterisk-Users] Codec G729 / x86_64 bits.   Dovid Bender
• [Asterisk-Users] Interesting Dial-Plan Question   Eric "ManxPower" Wieling
• [Asterisk-Users] Transfer - context/priority   Tomislav Parčina
• [Asterisk-Users] TE405P vs. SoundCard problem   Bob McDowell
• [Asterisk-Users] Interesting Dial-Plan Question   Matt
• [Asterisk-Users] Interesting Dial-Plan Question   Roshan Sembacuttiaratchy
• [Asterisk-Users] TE405P vs. SoundCard problem   Cory Andrews
• [Asterisk-Users] TE405P vs. SoundCard problem   Bob McDowell
• [Asterisk-Users] asterisk spandsp and txfax   Sarafoleanu Catalin
• [Asterisk-Users] Interesting Dial-Plan Question   Matt
• [Asterisk-Users] Unable to accept incoming PSTN calls   Time Bandit
• [Asterisk-Users] Interesting Dial-Plan Question   Eric "ManxPower" Wieling
• [Asterisk-Users] Codec G729 / x86_64 bits.   Alexander Lopez
• [Asterisk-Users] Seize phone line   Joe Pukepail
• [Asterisk-Users] Interesting Dial-Plan Question   Matt
• [Asterisk-Users] Camp on?   Andrew Kohlsmith
• [Asterisk-Users] PRI configuration   Wai Wu
• [Asterisk-Users] Seize phone line   Rich Adamson
• [Asterisk-Users] chan_sip.c patched with t.38   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Unable to accept incoming PSTN calls   Johnny Stork
• [Asterisk-Users] Guest Account - SIP and IAX   Brent Torrenga
• [Asterisk-Users] Unable to accept incoming PSTN calls   Johnny Stork
• [Asterisk-Users] Unable to accept incoming PSTN calls   Alex Robar
• [Asterisk-Users] Seize phone line   Time Bandit
• [Asterisk-Users] PRI configuration   Gareth Blades
• [Asterisk-Users] Unable to accept incoming PSTN calls   Time Bandit
• [Asterisk-Users] PRI configuration   Gareth Blades
• [Asterisk-Users] Very stupid question regarding Polycom Soundstation 4000   cristiangafotas
• [Asterisk-Users] Extreme delay before * processes call files   Jay Milk
• [Asterisk-Users] PRI configuration   Wai Wu
• [Asterisk-Users] Interesting Dial-Plan Question   Matt
• [Asterisk-Users] PRI configuration   Wai Wu
• [Asterisk-Users] Asterisk as a phone survey system   TV JOE
• [Asterisk-Users] Seize phone line   Joe Pukepail
• [Asterisk-Users] Slip/Frame Error between Mitel SX-200 and Asterisk   Geoff Manning
• [Asterisk-Users] Interesting Dial-Plan Question   Jay Milk
• [Asterisk-Users] some EICON Diva 4BRI questions   Klaus Darilion
• [Asterisk-Users] Seize phone line   Jay Milk
• [Asterisk-Users] Unable to accept incoming PSTN calls   Johnny Stork
• [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inbound call   Giorgio Incantalupo
• [Asterisk-Users] Asterisk as a phone survey system   Jay Milk
• [Asterisk-Users] access to caller/pickupgroup in extension.conf   Christoph Fürstaller
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] Unable to accept incoming PSTN calls   Time Bandit
• [Asterisk-Users] PRIs from two different telco   Wai Wu
• [Asterisk-Users] Polycom NTP issue   Kerry Garrison
• [Asterisk-Users] Early media after a dial command   hgaillac-sip at yahoo.fr
• [Asterisk-Users] astcc: need partial pin code   Ronald Wiplinger
• [Asterisk-Users] PRIs from two different telco   Matthew Fredrickson
• [Asterisk-Users] GrandStream GXP-2000   dataman
• [Asterisk-Users] PRIs from two different telco   C F
• [Asterisk-Users] Polycom NTP issue   Matt Florell
• [Asterisk-Users] Polycom NTP issue   Josué Conti
• [Asterisk-Users] Polycom NTP issue   Aaron Daniel
• [Asterisk-Users] Polycom NTP issue   Chris Mason (Lists)
• [Asterisk-Users] GrandStream GXP-2000   Michiel van Baak
• [Asterisk-Users] GrandStream GXP-2000   Waldo Rubinstein
• [Asterisk-Users] Receive fax (libtiff problem?)   jan.sarin at securia.se
• [Asterisk-Users] Polycom NTP issue   Kerry Garrison
• [Asterisk-Users] Polycom NTP issue   Kerry Garrison
• [Asterisk-Users] Polycom NTP issue   Aaron Daniel
• [Asterisk-Users] Quintum D3000   TJ Kells
• [Asterisk-Users] 7960G SIP Issue   sdcharly at gmail.com
• [Asterisk-Users] PRIs from two different telco   Steven Totaro
• [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inboundcall   Steven Totaro
• [Asterisk-Users] RE: GrandStream GXP-2000   Matthew Warren
• [Asterisk-Users] Analog GSM Gateways   GSM
• [Asterisk-Users] Polycom NTP issue   Aaron Daniel
• [Asterisk-Users] GrandStream GXP-2000   dataman
• [Asterisk-Users] Polycom NTP issue   Gary Reuter
• [Asterisk-Users] GrandStream GXP-2000   Mimmus
• [Asterisk-Users] Polycom NTP issue   Sean Cook
• [Asterisk-Users] PRIs from two different telco   Wai Wu
• [Asterisk-Users] Snom 320 HOLD and TRANSFER not detected   Tommaso Calosi
• [Asterisk-Users] Asterisk IVR / Scalability   Infobox Peru
• [Asterisk-Users] Recomended Commercial PBX Bundles/Software   Johnny Stork
• [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inboundcall   Eric "ManxPower" Wieling
• [Asterisk-Users] PRIs from two different telco   Eric "ManxPower" Wieling
• [Asterisk-Users] createlink option in agents.conf can't be disabled?   Kyle Sexton
• [Asterisk-Users] PrivacyManager & FastAGI: Rewrite or use?   Peter Beckman
• [Asterisk-Users] Polycom NTP issue   Rich Adamson
• [Asterisk-Users] Question on parkinglot   Mojo with Horan & Company, LLC
• [Asterisk-Users] PRIs from two different telco   Gary Reuter
• [Asterisk-Users] PRIs from two different telco   Matthew Fredrickson
• [Asterisk-Users] What happened to my subscription?   Martin Joseph
• [Asterisk-Users] Snom 320 HOLD and TRANSFER not detected   Franklin Webb
• [Asterisk-Users] PRIs from two different telco   Wai Wu
• [Asterisk-Users] FXO problems   Yenfry Nieves
• [Asterisk-Users] Extreme delay before * processes call files   Luki
• [Asterisk-Users] Polycom NTP issue   Kerry Garrison
• [Asterisk-Users] Looking for input on which way to go with small business setup   T. Shaw
• [Asterisk-Users] Polycom NTP issue   Chad Osmond
• [Asterisk-Users] Please Help i have a error with unicall and AT&T   Jorge Cisneros
• [Asterisk-Users] Looking for input on which way to go with smallbusiness setup   Chad Osmond
• [Asterisk-Users] Polycom NTP issue   Rich Adamson
• [Asterisk-Users] astcc: need partial pin code   Benchev
• [Asterisk-Users] Looking for input on which way to go with smallbusiness setup   Kerry Garrison
• [Asterisk-Users] Polycom NTP issue   Philippe Lindheimer
• [Asterisk-Users] Looking for input on which way to go with smallbusiness setup   Jean-Michel Hiver
• [Asterisk-Users] Polycom NTP issue   Kerry Garrison
• [Asterisk-Users] Looking for input on which way to gowith smallbusiness setup   Kerry Garrison
• [Asterisk-Users] Looking for input on which way to go with smallbusiness setup   Rich Adamson
• [Asterisk-Users] Analog GSM Gateways   John Novack
• [Asterisk-Users] SATA hard disk compatibility   John Novack
• [Asterisk-Users] Very stupid question regarding PolycomSoundstation 4000   Peter Johnson
• [Asterisk-Users] Looking for input on which way to go withsmallbusiness setup   T. Shaw
• [Asterisk-Users] USB conference phone   Steve Feinstein
• [Asterisk-Users] PRI got event: HDLC Bad FCS (8) on PrimaryD-channel of span   Anthony Rodgers
• [Asterisk-Users] Polycom 501 unregistered itself?   William M Conlon
• [Asterisk-Users] Looking for input on which way to go withsmallbusiness setup   Luki
• [Asterisk-Users] Asterisk as a phone survey system   Kris Cote
• [Asterisk-Users] TE405P vs. SoundCard problem   Leo Ann Boon
• [Asterisk-Users] Call Pickup with CID info   Bevan Blackie
• [Asterisk-Users] Looking for input on which way to gowithsmallbusiness setup   T. Shaw
• [Asterisk-Users] PRI got event: HDLC Bad FCS (8) on PrimaryD-channel of span   Eric "ManxPower" Wieling
• [Asterisk-Users] Info system   Ronald Wiplinger
• [Asterisk-Users] How can conference room can call out?   Ronald Wiplinger
• [Asterisk-Users] Polycom 501 unregistered itself?   Andrew Kohlsmith
• [Asterisk-Users] Camp on?   Eric "ManxPower" Wieling
• [Asterisk-Users] PRIs from two different telco   Andrew Kohlsmith
• [Asterisk-Users] PRIs from two different telco   Andrew Kohlsmith
• [Asterisk-Users] PRIs from two different telco   Wai Wu
• [Asterisk-Users] PRIs from two different telco   Wai Wu
• [Asterisk-Users] PRIs from two different telco   Wai Wu
• [Asterisk-Users] Camp on?   Nathan Alberti
• [Asterisk-Users] HINTING, how it works... Please explain   Asterisk
• [Asterisk-Users] USB conference phone   John covici
• [Asterisk-Users] billing realtime   JP Carballo
• [Asterisk-Users] 407 proxy authentication   hgaillac-sip at yahoo.fr
• [Asterisk-Users] astcc: need partial pin code   JP Carballo
• [Asterisk-Users] CHANUNAVAIL, busy and congestion   Mark Edwards
• [Asterisk-Users] astcc: need partial pin code   JP Carballo
• [Asterisk-Users] astcc: need partial pin code   JP Carballo
• [Asterisk-Users] Asterisk as a phone survey system   JP Carballo
• [Asterisk-Users] Very stupid question regarding PolycomSoundstation 4000   cristiangafotas
• [Asterisk-Users] astcc: need partial pin code   Benchev
• [Asterisk-Users] USB conference phone   Kerry Garrison
• [Asterisk-Users] some EICON Diva 4BRI questions   Klaus Darilion
• [Asterisk-Users] mISDN: No DID/extension information returns busy to caller   Ralf Schlatterbeck
• [Asterisk-Users] Help on multiple dialed sip-channels.   Arne Morten Johansen
• [Asterisk-Users] Asterisk dialing   Andrew Nowrot
• [Asterisk-Users] (no subject)   Soner Tari
• [Asterisk-Users] how to dial the real time iax user   陈帆
• [Asterisk-Users] Re: Extreme delay before * processes call files   Remco Barende
• [Asterisk-Users] Which h323 channel for asterisk and gnugk ?   hgaillac-sip at yahoo.fr
• [Asterisk-Users] video support on iax trunk   asterisk at frameweb.it
• [Asterisk-Users] caching of sip account   Mimmus
• [Asterisk-Users] mISDN: No DID/extension information returns busy to caller   Dmitry Ivanov
• [Asterisk-Users] USB conference phone   Steve Feinstein
• [Asterisk-Users] IVR answers and questions instead of MOH in a queue, how?   Marco Mouta
• [Asterisk-Users] Camp on?   Eric "ManxPower" Wieling
• [Asterisk-Users] asterisk spandsp and txfax   Alejandro Vargas
• [Asterisk-Users] USB conference phone   John covici
• [Asterisk-Users] features.conf   José Luis Gómez
• [Asterisk-Users] Re: Extreme delay before * processes call files   Doug Lytle
• [Asterisk-Users] Dial 'R' option gone?   Benoit Panizzon
• [Asterisk-Users] How to transfer outgoing calls   Hans-Peter Straub
• [Asterisk-Users] How to transfer outgoing calls   Time Bandit
• [Asterisk-Users] some EICON Diva 4BRI questions   David Waugh
• [Asterisk-Users] asterisk spandsp and txfax   Sarafoleanu Catalin
• [Asterisk-Users] Dial 'R' option gone?   Eric "ManxPower" Wieling
• [Asterisk-Users] How to transfer outgoing calls   Eric "ManxPower" Wieling
• [Asterisk-Users] PRIs from two different telco   Don Pobanz
• [Asterisk-Users] Integrics release Enswitch 2.0   Alistair Cunningham
• [Asterisk-Users] caching of sip account   Aaron Daniel
• [Asterisk-Users] PRIs from two different telco   Wai Wu
• [Asterisk-Users] Integrics release Enswitch 2.0   Joshua Colp
• R: [Asterisk-Users] Snom 320 HOLD and TRANSFER not detected   Tommaso Calosi
• [Asterisk-Users] Intercom Phones and Asterisk   Johnny Stork
• [Asterisk-Users] Warning: No path to translate with SJPhone   Carlos Alberto Bernat Orozco
• [Asterisk-Users] How to transfer outgoing calls   Hans-Peter Straub
• [Asterisk-Users] PRIs from two different telco   Andrew Kohlsmith
• [Asterisk-Users] Some Extensions Remain Busy?   Johnny Stork
• [Asterisk-Users] Grandstream GXP-2000   Johnny Stork
• [Asterisk-Users] TE405P vs. SoundCard problem   Bob McDowell
• [Asterisk-Users] caching of sip account   Mimmus
• [Asterisk-Users] Explain to me VoIP termination service.   Matt
• [Asterisk-Users] Grandstream GXP-2000   Gareth Blades
• [Asterisk-Users] 482 Loop Detected on sip calls   Ajit
• [Asterisk-Users] Grandstream GXP-2000   Waldo Rubinstein
• [Asterisk-Users] Integrics release Enswitch 2.0   Chris Mason (Lists)
• [Asterisk-Users] RESOLVED - TE405P vs. SoundCard problem (in reality - TE405P No Voice Problem)   Bob McDowell
• [Asterisk-Users] Some Extensions Remain Busy?   Marco Mouta
• [Asterisk-Users] caching of sip account   James Nunnerley
• [Asterisk-Users] features.conf   Ronald Wiplinger
• [Asterisk-Users] 482 Loop Detected on sip calls   Joshua Colp
• [Asterisk-Users] Problems if GXP-2000 phones and Asterisk are not on the same network   Mimmus
• [Asterisk-Users] RESOLVED - TE405P vs. SoundCard problem (in reality - TE405P No Voice Problem)   Rich Adamson
• [Asterisk-Users] Asterisk dialing   Time Bandit
• [Asterisk-Users] Some Extensions Remain Busy?   Johnny Stork
• [Asterisk-Users] Grandstream GXP-2000   Johnny Stork
• [Asterisk-Users] Warning: No path to translate with SJPhone   Eric "ManxPower" Wieling
• [Asterisk-Users] Interesting Dial-Plan Question   Jonathan k. Creasy
• [Asterisk-Users] Basic Linux Advice   Thczv F. Thczv
• [Asterisk-Users] features.conf   From PH
• [Asterisk-Users] Interesting Dial-Plan Question   Eric "ManxPower" Wieling
• [Asterisk-Users] codec preference   From PH
• [Asterisk-Users] Basic Linux Advice   Kristian Kielhofner
• [Asterisk-Users] OT: Phishing with phones   Andrew Latham
• [Asterisk-Users] Intercom Phones and Asterisk   Ira
• [Asterisk-Users] New astGUIclient VICIDIAL Release 1.1.11   Matt Florell
• [Asterisk-Users] What is i2 ? 911 Candian Style   Bob's Leaky News Service
• [Asterisk-Users] FW: NuFone Update: DIDs   Matt
• [Asterisk-Users] Official TE411P echo settings??   Steven
• [Asterisk-Users] FW: NuFone Update: DIDs   Tom Vile
• [Asterisk-Users] Re: Slip/Frame Error between Mitel SX-200 and Asterisk   Geoff Manning
• [Asterisk-Users] Dual Timing Sources   Matt
• [Asterisk-Users] FW: NuFone Update: DIDs   Matt
• [Asterisk-Users] FW: NuFone Update: DIDs   Rich Adamson
• [Asterisk-Users] Dual Timing Sources   Andrew Kohlsmith
• [Asterisk-Users] Dual Timing Sources   Steve Underwood
• [Asterisk-Users] Dual Timing Sources   Eric "ManxPower" Wieling
• [Asterisk-Users] Configuration OpenPri for logger   Andres Gomez
• [Asterisk-Users] Odd internal vs. External dialplan issue   Steven
• [Asterisk-Users] Official TE411P echo settings??   Rob Lith
• [Asterisk-Users] Dual Timing Sources   Matt
• [Asterisk-Users] Re: Official TE411P echo settings??   Steven
• [Asterisk-Users] DNSMasq - Why the stuff hits the fan when the net connection is down   Brent Torrenga
• [Asterisk-Users] Re: Official TE411P echo settings??   Rob Lith
• [Asterisk-Users] Digium TE210P and faxing, is it possible?   Dan Brummer
• [Asterisk-Users] Asterisk and Panasonic KX-T336   Carlos Rojas
• [Asterisk-Users] Asterisk FAx   Wasif
• [Asterisk-Users] Cell phones and DTMF   Bob McDowell
• [Asterisk-Users] RE: Cell phones and DTMF   Bob McDowell
• [Asterisk-Users] Bristuff 1.2.7.1?   Vidar
• [Asterisk-Users] FW: NuFone Update: DIDs   Chris Mason (Lists)
• [Asterisk-Users] Asterisk DNID/RDNIS with Dial iax2   Andrew
• [Asterisk-Users] How to use the cmd SMS   Il Neofita
• [Asterisk-Users] FW: NuFone Update: DIDs   Lacy Moore - Aspendora
• [Asterisk-Users] Asterisk Hangs the whole system   pdhales at optusnet.com.au
• [Asterisk-Users] Rhino T1 and 4-port FXO cards   Leo Ann Boon
• [Asterisk-Users] Problems if GXP-2000 phones and Asterisk are not on the same network   Waldo Rubinstein
• [Asterisk-Users] two box share one real time configuration database.   陈帆
• [Asterisk-Users] Asterisk DNID/RDNIS with Dial iax2   Steve Totaro
• [Asterisk-Users] two box share one real time configuration database.   billy at kersting.com
• [Asterisk-Users] Remote UNIX connection disconnected over and over   Il Neofita
• [Asterisk-Users] two box share one real time configuration database.   陈帆
• [Asterisk-Users] Call Queue Transfer   Josué Conti
• [Asterisk-Users] Remote UNIX connection disconnected over and over   JP Carballo
• [Asterisk-Users] Random 1-way audio on IAX2 Connections   Aryn Nakaoka
• [Asterisk-Users] Random 1-way audio on IAX2 Connections   billy at kersting.com
• [Asterisk-Users] [SPAM] Re: [asterisk-dev] Announcing Astmanproxy 1.20   Dinesh Nair
• [Asterisk-Users] Disable 407 proxy authentication for outbound domains   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Dual Timing Sources   qrss at keitz.org
• [Asterisk-Users] FW: NuFone Update: DIDs   Kerry Garrison
• [Asterisk-Users] IAX + GSM codec is good quality   陈帆
• [Asterisk-Users] Asterisk DNID/RDNIS with Dial iax2   Alexander Lopez
• [Asterisk-Users] IAX + GSM codec is good quality   Josué Conti
• [Asterisk-Users] stupid trick of the day (fried polycom)   asterisk at txpe.net
• [Asterisk-Users] Dial 'R' option gone?   Benoit Panizzon
• [Asterisk-Users] Call Queue Transfer   Dinesh Nair
• [Asterisk-Users] Is there a way to monitor the DTMF tones on a channel?   Obelix
• [Asterisk-Users] stupid trick of the day (fried polycom)   Kevin P. Fleming
• [Asterisk-Users] Help with Mediatrix 1204   Frank Attard
• [Asterisk-Users] NOTIFY Problem   Il Neofita
• [Asterisk-Users] Help with Mediatrix 1204   Kevin P. Fleming
• [Asterisk-Users] Dial 'R' option gone?   Doug Lytle
• [Asterisk-Users] Call Queue Transfer   Josué Conti
• [Asterisk-Users] NOTIFY Problem   tom
• [Asterisk-Users] asterisk to use an outbound proxy   Raymond Chen
• [Asterisk-Users] asterisk to use an outbound proxy   Kevin P. Fleming
• [Asterisk-Users] some EICON Diva 4BRI questions   Armin Schindler
• [Asterisk-Users] FW: NuFone Update: DIDs   Matt
• [Asterisk-Users] Dual Timing Sources   Matt
• [Asterisk-Users] Dial 'R' option gone?   Eric "ManxPower" Wieling
• [Asterisk-Users] canreinvite, bandwidth, dial option   Ronald Wiplinger
• [Asterisk-Users] Telephone support charging system with Asterisk?   Mike Dent
• [Asterisk-Users] Problems if GXP-2000 phones and Asterisk are not on the same network   Faris Raouf
• [Asterisk-Users] How many asterisk process's are "normal"?   T.S
• [Asterisk-Users] How many asterisk process's are "normal"?   Joshua Colp
• [Asterisk-Users] How many asterisk process's are "normal"?   Doug Lytle
• [Asterisk-Users] Locate Me Function with freePBX   Kerry Garrison
• [Asterisk-Users] NOTIFY Problem   Il Neofita
• [Asterisk-Users] Telephone support charging system with Asterisk?   JP Carballo
• [Asterisk-Users] Audio Muting at seemingly random times   Matthew Drobnak
• [Asterisk-Users] Asterisk dialing   Andrew Nowrot
• [Asterisk-Users] USB conference phone   mgraves at mstvp.com
• [Asterisk-Users] Unable to Make Asterisk-addons   Dan Journo
• [Asterisk-Users] Large Asterisk with Regexten, Regcontext, DUNDi, , , , , , , , , but not load balance...   陈帆
• [Asterisk-Users] RE: Install/Upgrade   Dave Morrow
• [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk   Steve Totaro
• [Asterisk-Users] RE: Install/Upgrade   Steve Totaro
• [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk   Alexander Lopez
• [Asterisk-Users] FW: NuFone Update: DIDs   Steve Totaro
• [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk   Steve Totaro
• [Asterisk-Users] RE: Install/Upgrade   Kerry Garrison
• [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk   Ira
• [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk   Steve Totaro
• [Asterisk-Users] Codec G729 no longer works.   Jason A. Kates
• [Asterisk-Users] Codec G729 no longer works.   Joshua Colp
• [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk   Paul Dugas
• [Asterisk-Users] [OT]Cisco 2621XM with (2) T1/PRI inetrfaces for sale   Dan Austin
• [Asterisk-Users] Codec G729 no longer works.   Jason A. Kates
• [Asterisk-Users] Re: Large Asterisk with Regexten, Regcontext, DUNDi, , , , , , , , , but not load balance...   陈帆
• [Asterisk-Users] Compare to Skype   Ronald Wiplinger
• [Asterisk-Users] NuFone - How to switch to another provider?   Ronald Wiplinger
• [Asterisk-Users] Compare to Skype   Gabriel Afana
• [Asterisk-Users] Compare to Skype   Greg Oliver
• [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk   Kristian Kielhofner
• [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk   Kristian Kielhofner
• [Asterisk-Users] Re: Compare to Skype   Yaakov Menken
• [Asterisk-Users] problame with outbound calls on pri   Doug Langley
• [Asterisk-Users] Re: Trial Version of Asterisk Interface Available   Vikram Rangnekar
• [Asterisk-Users] PRI got event: HDLC Bad FCS (8) on PrimaryD-channel of span   Nico Giefing
• [Asterisk-Users] Digium cards, so disappointing !   Boris Bakchiev
• [Asterisk-Users] problame with outbound calls on pri   Steve Totaro
• [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk   Steve Totaro
• [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk   Peter Bowyer
• [Asterisk-Users] How to monitor DTMF tones in a call?   Obelix
• [Asterisk-Users] FW: NuFone Update: DIDs   Wilson Pickett
• [Asterisk-Users] stupid trick of the day (fried polycom)   Wilson Pickett
• [Asterisk-Users] Error : ast_readaudio_callback: Failed to write frame   Hatami Nugraha
• [Asterisk-Users] stupid trick of the day (fried polycom)   Bruce Reeves
• [Asterisk-Users] some sip clients unreachable on sip-reload   Florian Meister
• [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk   Assaf Flatto
• [Asterisk-Users] Fwd: can modify CHAN_SIP.c to generate a new exten=> ext, 2, dial(tech/peer) ?   陈帆
• [Asterisk-Users] Error : ast_readaudio_callback: Failed to write frame   Imran Ahmed
• [Asterisk-Users] Compare to Skype   mgraves at mstvp.com
• [Asterisk-Users] Compare to Skype   Ronald Wiplinger
• [Asterisk-Users] Compare to Skype   mgraves at mstvp.com
• [Asterisk-Users] Compare to Skype   Ronald Wiplinger
• [Asterisk-Users] Intermittent problem dialling out on a SIP channel   hugolivude
• [Asterisk-Users] Compare to Skype   Eric "ManxPower" Wieling
• [Asterisk-Users] problame with outbound calls on pri   Eric "ManxPower" Wieling
• [Asterisk-Users] newbie-too much latency   Ryder Brook
• [Asterisk-Users] Legacy PBX integration   Remco Barende
• [Asterisk-Users] problame with outbound calls on pri   Doug Langley
• [Asterisk-Users] Change in audio file while listening to it   Marco Trucchi
• [Asterisk-Users] Asterisk is stripping my area code   Jim Lynch
• [Asterisk-Users] RE: Asterisk is stripping my area code   Jim Lynch
• [Asterisk-Users] Codec G729 no longer works.   Patrick
• [Asterisk-Users] RE: Asterisk is stripping my area code   Kerry Garrison
• [Asterisk-Users] Codec G729 no longer works.   Mathieu Chouquet-Stringer
• [Asterisk-Users] RE: Asterisk is stripping my area code   Jim Lynch
• [Asterisk-Users] RE: Asterisk is stripping my area code   Kerry Garrison
• [Asterisk-Users] Asterisk 1.2.7.1 & Fritz!PCI or AVM A1   Rainer Maier
• [Asterisk-Users] Asterisk 1.2.7.1 & Fritz!PCI or AVM A1   Rainer Maier
• [Asterisk-Users] Compare to Skype   Time Bandit
• [Asterisk-Users] newbie-too much latency   Time Bandit
• [Asterisk-Users] integrated voip originator, to digitize audio once and only once?   Tom Engleward
• [Asterisk-Users] Digium cards, so disappointing !   Remco Barende
• [Asterisk-Users] RE: Asterisk is stripping my area code   Jim Lynch
• [Asterisk-Users] FreePBX in production?   Rich Adamson
• [Asterisk-Users] FreePBX in production?   Avi Miller
• [Asterisk-Users] FreePBX in production?   Kerry Garrison
• [Asterisk-Users] FreePBX in production?   Ariel Batista
• [Asterisk-Users] How to monitor DTMF tones in a call?   Moises Silva
• [Asterisk-Users] FreePBX in production?   Rich Adamson
• [Asterisk-Users] FreePBX in production?   Avi Miller
• [Asterisk-Users] FW: NuFone Update: DIDs   Hermann Wecke
• [Asterisk-Users] FreePBX in production?   Rich Adamson
• [Asterisk-Users] FreePBX in production?   Avi Miller
• [Asterisk-Users] Legacy PBX integration   Jerry Jones
• [Asterisk-Users] Asterisk is stripping my area code   Jerry Jones
• [Asterisk-Users] FreePBX in production?   Steve Totaro
• [Asterisk-Users] FW: NuFone Update: DIDs   Steve Totaro
• [Asterisk-Users] FreePBX in production?   Time Bandit
• [Asterisk-Users] FreePBX in production?   Rich Adamson
• [Asterisk-Users] WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '????'   hugolivude
• [Asterisk-Users] FreePBX in production?   Rich Adamson
• [Asterisk-Users] FreePBX in production?   Avi Miller
• [Asterisk-Users] FreePBX in production?   Avi Miller
• [Asterisk-Users] Legacy PBX integration   Lacy Moore - Aspendora
• [Asterisk-Users] FreePBX in production?   Rich Adamson
• [Asterisk-Users] FreePBX in production?   Avi Miller
• [Asterisk-Users] queues   Patrick Siglin
• [Asterisk-Users] queues   Kerry Garrison
• [Asterisk-Users] Legacy PBX integration   Remco Barende
• [Asterisk-Users] Legacy PBX integration   Remco Barende
• [Asterisk-Users] Legacy PBX integration   Lacy Moore - Aspendora
• [Asterisk-Users] Legacy PBX integration   Remco Barende
• [Asterisk-Users] SATA hard disk compatibility   amna saleem
• [Asterisk-Users] PRI Issue: D-Channel woes   Terence Burnard
• [Asterisk-Users] Compare to Skype   Matt Ranney
• [Asterisk-Users] Call Queue Transfer   Dinesh Nair
• [Asterisk-Users] Legacy PBX integration   Lacy Moore - Aspendora
• [Asterisk-Users] how to make messages button on ip500 work   PCSUPPORT PC SUPPORT