June 2007 Archives by date

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Starting: Fri Jun 1 02:15:22 CDT 2007
Ending: Sat Jun 30 23:10:43 CDT 2007
Messages: 2023

• [asterisk-users] False ring problem   Rizwan Hisham
• [asterisk-users] moh backround?   Thomas Stein
• [asterisk-users] moh backround?   Thomas Stein
• [asterisk-users] Context documentation for the newbie!   Mats Karlsson
• [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO!   Jaswinder Singh
• [asterisk-users] Cutted audio or 2/3s blanks on EuroISDN- Asterisk1.4   Steve Hanselman
• [asterisk-users] ZAP inbound/outbound connection taking too long   Gavin Henry
• [asterisk-users] Net2Phone Multiple SIP Trunk Not Working   Jaswinder Singh
• [asterisk-users] CARD FOR inband signal   Tzafrir Cohen
• [asterisk-users] Audio going blank for a few seconds and thencomes back. What could be the reason?   Steve Hanselman
• [asterisk-users] how can qualify=yes trigger some external event?   Ricardo Carvalho
• [asterisk-users] Net2Phone Multiple SIP Trunk Not Working   Salah Eddine ELMRABET
• [asterisk-users] ZAP inbound/outbound connection taking too long   Gordon Henderson
• [asterisk-users] High Port Count ATA   Jerry Jones
• [asterisk-users] Audio going blank for a few seconds and thencomes back. What could be the reason?   Zeeshan Zakaria
• [asterisk-users] OT Slightly:   Dean Collins
• [asterisk-users] how can qualify=yes trigger some external event?   Watkins, Bradley
• [asterisk-users] Audio going blank for a few seconds andthencomes back. What could be the reason?   Steve Hanselman
• [asterisk-users] reset Polycom phones remotely   Rob Schall
• [asterisk-users] Thank you Asterisk mailing list!   Ricardo Martins
• [asterisk-users] Audio going blank for a few seconds andthencomes back. What could be the reason?   Rob Schall
• max verbose level [was: Re: [asterisk-users] ZAP inbound/outbound connection taking too long]   Tzafrir Cohen
• [asterisk-users] High Port Count ATA   Natambu Obleton
• max verbose level [was: Re: [asterisk-users] ZAP inbound/outbound connection taking too long]   Gordon Henderson
• [asterisk-users] Audio going blank for a few seconds andthencomes back. What could be the reason?   Zeeshan Zakaria
• [asterisk-users] Audio going blank for a few seconds andthencomes back. What could be the reason?   Andrew Kohlsmith
• [asterisk-users] RF to IP bridge   John Treble
• [asterisk-users] SIP & NAT ...   Gordon Henderson
• OT: "The Ignorance of Crowds" (was: [asterisk-users] OT Slightly: )   Matthew Rubenstein
• [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes   John Hughes
• [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes   John Hughes
• [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks   John Hughes
• [asterisk-users] False ring problem   Eric "ManxPower" Wieling
• [asterisk-users] Audio going blank for a few seconds andthencomesback. What could be the reason?   Steve Hanselman
• [asterisk-users] G729 client and server Side   ram
• [asterisk-users] Asteris et winsip   khawla khawla
• [asterisk-users] SIP & NAT ...   Anthony Francis
• [asterisk-users] Meetme problems   ram
• [asterisk-users] Asteris et winsip   khawla khawla
• [asterisk-users] how can qualify=yes trigger some external event?   Anthony Francis
• [asterisk-users] False ring problem   Ricardo Martins
• [asterisk-users] multiple host= in sip.conf   Anthony Francis
• [asterisk-users] asterisk mysql support   Diego Quintana Cruz
• [asterisk-users] how can qualify=yes trigger some external event?   Watkins, Bradley
• [asterisk-users] asterisk mysql support   David Ruggles
• [asterisk-users] chan_iax2.so issues   Simon Alman
• [asterisk-users] Cisco 7961G   Eric Lubow
• [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes   Matthew J. Roth
• [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes - Low volume benchmarks   Matthew J. Roth
• [asterisk-users] SIP & NAT ...   Tom Rymes
• [asterisk-users] SIP & NAT ...   Gordon Henderson
• [asterisk-users] Audio going blank for a few seconds andthencomes back. What could be the reason?   Rob Schall
• [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yieldinggains at high call volumes   Douglas Garstang
• [asterisk-users] asterisk mysql support   Tzafrir Cohen
• [asterisk-users] how can qualify=yes trigger some external event?   rjcarvalho at reit.up.pt
• [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes   Stephen Davies
• [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yieldinggains at high call volumes   Stephen Davies
• [asterisk-users] asterisk mysql support   Douglas Garstang
• [asterisk-users] Cutted audio or 2/3s blanks on EuroISDN- Asterisk1.4   Matthew Fredrickson
• [asterisk-users] applicationmap on features   Carlos Chavez
• [asterisk-users] Asterisk and CCM 5.x SIP trunk   Greg Oliver
• [asterisk-users] asterisk mysql support   Tzafrir Cohen
• [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes   Matthew J. Roth
• [asterisk-users] how can qualify=yes trigger some external event?   Anthony Francis
• [asterisk-users] reset Polycom phones remotely   Mojo with Horan & Company, LLC
• [asterisk-users] Asterisk Time Card   Mojo with Horan & Company, LLC
• [asterisk-users] Cisco 7961G   Pavel Jezek
• [asterisk-users] Audio going blank for a few seconds and then comes back. What could be the reason?   Matthew Fredrickson
• [asterisk-users] Audio going blank for a few seconds andthencomesback. What could be the reason?   Matthew Fredrickson
• [asterisk-users] SIP & NAT ...   Gordon Henderson
• [asterisk-users] Cisco 7961G   Greg Oliver
• [asterisk-users] SugarCRM Integration   Diego Quintana Cruz
• [asterisk-users] Re: OpenVox A400P01on thin client?   Vincent
• [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING   Alvaro Parres
• [asterisk-users] ZAP inbound/outbound connection taking too long   Gavin Henry
• [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes   Stephen Davies
• [asterisk-users] Cisco 7961G   Eric Lubow
• [asterisk-users] ZAP inbound/outbound connection taking too long   Drew Gibson
• [asterisk-users] Gateway VIP450FO and VIP 400FO   MCelo
• [asterisk-users] OutBound dial plan   Eddy Pimntel
• [asterisk-users] Call Back Service   Costa Dinoteli
• [asterisk-users] WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 36458 udptl t38   ram
• [asterisk-users] SugarCRM Integration   Joseph Bajin
• [asterisk-users] click to call   Anton Krall
• [asterisk-users] SugarCRM Integration   Crazy Boy
• [asterisk-users] click to call   Gordon Henderson
• [asterisk-users] False ring problem   Rizwan Hisham
• [asterisk-users] Cisco 7961G   Pavel Jezek
• [asterisk-users] SugarCRM Integration   Duncan Turnbull
• [asterisk-users] Asterisk registering problem   we at deuromedia.at
• [asterisk-users] Asterisk registering problem   Anselm Martin Hoffmeister
• [asterisk-users] RF to IP bridge   F6HQZ
• [asterisk-users] SugarCRM Integration   Chris Mason (Lists)
• [asterisk-users] click to call   Steve Totaro
• [asterisk-users] Cutted audio or 2/3s blanks on EuroISDN- Asterisk1.4   Administrator TOOTAI
• [asterisk-users] click to call   Steve Totaro
• [asterisk-users] click to call   Joseph Bajin
• [asterisk-users] how to make busy sign   Csaba Tóth
• [asterisk-users] linksys pap2 version2 ata DTMF issue   Troy Ayers
• [asterisk-users] click to call   Steve Totaro
• [asterisk-users] click to call   Dean Collins
• [asterisk-users] System Application, Fail/Timeout Issue   thedss at ntlworld.com
• [asterisk-users] how to make busy sign   Steve Totaro
• [asterisk-users] Dynamically adding Context in dialplan?   Nasir Iqbal
• [asterisk-users] linksys wip300 firmware   Ilan Rabinovitch
• [asterisk-users] click to call   Nick Seraphin
• [asterisk-users] click to call   Nick Seraphin
• [asterisk-users] click to call   Dean Collins
• [asterisk-users] click to call   Steve Totaro
• [asterisk-users] G729 client and server Side   Bruno
• [asterisk-users] click to call   Dean Collins
• [asterisk-users] really strange behavior   BSumrall
• [asterisk-users] really strange behavior   Jared Smith
• [asterisk-users] click to call   Steve Totaro
• [asterisk-users] click to call   Al Bochter
• [asterisk-users] Call Back Service   Lee Jenkins
• [asterisk-users] System Application, Fail/Timeout Issue   Lee Jenkins
• [asterisk-users] Auto Dial Problem   Lee Jenkins
• [asterisk-users] click to call   Nick Seraphin
• [asterisk-users] Auto Dial Problem   Steve Totaro
• [asterisk-users] Passing call duration to an AGI Script   Adi Simon
• [asterisk-users] Asterisk Crash   Arun Kumar
• [asterisk-users] really strange behavior   randulo
• [asterisk-users] Asterisk Queue   Arun Kumar
• [asterisk-users] Auto Dial Problem   Nasir Iqbal
• [asterisk-users] Auto Dial Problem   Nasir Iqbal
• [asterisk-users] Asterisk Queue   Mattt
• [asterisk-users] Centos kernel source   Khaled Chehab
• [asterisk-users] Asterisk Queue   Arun Kumar
• [asterisk-users] Asterisk Queue   Arun Kumar
• [asterisk-users] how to make busy sign   Tóth Csaba
• [asterisk-users] zaptel on CENTOS servercd   Khaled Chehab
• [asterisk-users] Telefonica in Czech Republic Blocking VOIP ?   Dovid B
• [asterisk-users] linksys wip300 firmware   Bryan Laird
• FW: [asterisk-users] Centos kernel source   Khaled Chehab
• FW: [asterisk-users] Centos kernel source   Khaled Chehab
• [asterisk-users] System Application, Fail/Timeout Issue   thedss at ntlworld.com
• [asterisk-users] Centos kernel source   Tzafrir Cohen
• [asterisk-users] Centos kernel source   Steve Totaro
• [asterisk-users] zaptel on CENTOS servercd   Steve Totaro
• [asterisk-users] zaptel on CENTOS servercd   Tzafrir Cohen
• [asterisk-users] Centos kernel source   Philipp Kempgen
• [asterisk-users] zaptel on CENTOS servercd   Steve Totaro
• [asterisk-users] Centos kernel source   Steve Totaro
• [asterisk-users] zaptel on CENTOS servercd   Tzafrir Cohen
• FW: [asterisk-users] Centos kernel source   Tzafrir Cohen
• [asterisk-users] Centos kernel source   Steve Totaro
• [asterisk-users] Centos kernel source   Tzafrir Cohen
• [asterisk-users] really strange behavior   BSumrall
• [asterisk-users] Centos kernel source   Philipp Kempgen
• [asterisk-users] reset Polycom phones remotely   Stephen Bosch
• [asterisk-users] asterisk auto dial does not wait for answer   Stephen Bosch
• [asterisk-users] Thank you Asterisk mailing list!   Stephen Bosch
• [asterisk-users] ZAP inbound/outbound connection taking too long   Stephen Bosch
• [asterisk-users] Loud noise instead of MOH   Mauro Zanin
• [asterisk-users] Chan_mobile issue   Steve Totaro
• [asterisk-users] really strange behavior   Tom Lynn
• [asterisk-users] Loud noise instead of MOH   Gang Chen
• [asterisk-users] SIP Options Reply Ignored   Ian Clough
• [asterisk-users] Strange problem with channel allocation   Jonson Player
• [asterisk-users] wifi sip phone real-world experiences?   Andrew Kohlsmith
• [asterisk-users] SIP Options Reply Ignored   Alex Balashov
• [asterisk-users] asterisk auto dial does not wait for answer   Eric "ManxPower" Wieling
• [asterisk-users] really strange behavior   Eric "ManxPower" Wieling
• [asterisk-users] really strange behavior   Eric "ManxPower" Wieling
• [asterisk-users] wifi sip phone real-world experiences?   Alex Crow
• [asterisk-users] wifi sip phone real-world experiences?   Gordon Henderson
• [asterisk-users] zaptel on CENTOS servercd   Lee Jenkins
• [asterisk-users] wifi sip phone real-world experiences?   Andrew Kohlsmith
• [asterisk-users] Auto Dial Problem   Lee Jenkins
• [asterisk-users] Can two card be configured on same machine.   Sanjay Rajdev
• [asterisk-users] linksys wip300 firmware   Ilan Rabinovitch
• [asterisk-users] Can two card be configured on same machine.   John Novack
• [asterisk-users] click to call   Anton Krall
• [asterisk-users] click to call   Anton Krall
• [asterisk-users] click to call   Anton Krall
• [asterisk-users] Chan_mobile issue   Jared Bellows
• [asterisk-users] SIP Options Reply Ignored   Ian Clough
• [asterisk-users] click to call   Tzafrir Cohen
• [asterisk-users] zaptel on CENTOS servercd   Tzafrir Cohen
• [asterisk-users] nvlinedetect for Asterisk 1.4   aslay at pinwee.com.my
• [asterisk-users] yum om centos   Khaled Chehab
• [asterisk-users] click to call   Tim Panton
• [asterisk-users] click to call   Tim Panton
• [asterisk-users] Mixing Vars into Voicemail WAVs   Robert Goodyear
• [asterisk-users] wifi sip phone real-world experiences?   Tim Panton
• [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING   Tobias Wolf
• [asterisk-users] Digium Card   Arun Kumar
• [asterisk-users] Issue with Grandstream ATA 496.   Mauro Zanin
• [asterisk-users] Wireless IP Phone with external Telephone Book   Tobias Wolf
• [asterisk-users] Digium Card   Tzafrir Cohen
• [asterisk-users] Cutted audio or 2/3s blankson EuroISDN- Asterisk1.4   Steve Hanselman
• [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING   Humberto Figuera
• [asterisk-users] click to call   Gordon Henderson
• [asterisk-users] wifi sip phone real-world experiences?   Gordon Henderson
• [asterisk-users] yum om centos   Jaswinder Singh
• [asterisk-users] Digium Card   Andrew Latham
• [asterisk-users] Re: click to call   Steven
• [asterisk-users] yum om centos   Khaled Chehab
• [asterisk-users] wifi sip phone real-world experiences?   Bryan Laird
• [asterisk-users] IAX2 Trunk Problem   Arun Kumar
• [asterisk-users] zaptel on CENTOS servercd   Jim Suber
• [asterisk-users] wifi sip phone real-world experiences?   Andrew Kohlsmith
• [asterisk-users] Wireless IP Phone with external Telephone Book   Dave Bour
• [asterisk-users] G729 License   Arun Kumar
• [asterisk-users] wifi sip phone real-world experiences?   Bryan Laird
• [asterisk-users] debug logs   ram
• [asterisk-users] G729 License   olivier.taylor
• [asterisk-users] zaptel on CENTOS servercd   John Novack
• [asterisk-users] G729 License   ram
• [asterisk-users] CDR timing   Steve Murphy
• [asterisk-users] zaptel on CENTOS servercd   Lee Jenkins
• [asterisk-users] wifi sip phone real-world experiences?   Zoa
• [asterisk-users] Loud noise instead of MOH   Lee Jenkins
• [asterisk-users] Wireless IP Phone with external Telephone Book   Gordon Henderson
• [asterisk-users] CDR timing   Rob Schall
• [asterisk-users] wifi sip phone real-world experiences?   Alban
• [asterisk-users] addqueuemember recording and reporting problems   Jordan Novak
• [asterisk-users] Re: click to call   Anton Krall
• [asterisk-users] wifi sip phone real-world experiences?   Paul Hayes
• [asterisk-users] FX Dialing Odd   Rob Schall
• [asterisk-users] background dialing   Thomas Stein
• [asterisk-users] any codec passthru mode   Jaswinder Singh
• [asterisk-users] FX Dialing Odd   John Novack
• [asterisk-users] wifi sip phone real-world experiences?   Andrew Kohlsmith
• [asterisk-users] ringback detection   Jaswinder Singh
• [asterisk-users] any codec passthru mode   SIP
• [asterisk-users] wifi sip phone real-world experiences?   Alex Crow
• [asterisk-users] debug logs   Tzafrir Cohen
• [asterisk-users] FX Dialing Odd   Rob Schall
• [asterisk-users] no dtmf pcom 650 only outbound calls   A_ Navone
• [asterisk-users] any codec passthru mode   Pavel Jezek
• [asterisk-users] FX Dialing Odd   John Novack
• [asterisk-users] any codec passthru mode   Rizwan Hisham
• [asterisk-users] FX Dialing Odd   Rob Schall
• [asterisk-users] wifi sip phone real-world experiences?   Zoa
• [asterisk-users] wifi sip phone real-world experiences?   Alban
• [asterisk-users] FX Dialing Odd   John Novack
• [asterisk-users] FX Dialing Odd   Eric "ManxPower" Wieling
• [asterisk-users] wifi sip phone real-world experiences?   olivier.taylor
• [asterisk-users] AEL2 Includes in Macro...   Douglas Garstang
• [asterisk-users] FX Dialing Odd   David Boyd
• [asterisk-users] Compilation after Source code changes in Asterisk   Mojo with Horan & Company, LLC
• [asterisk-users] FX Dialing Odd   Eric "ManxPower" Wieling
• [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING   Tobias Wolf
• [asterisk-users] Debug meetme   Adrian Marsh
• [asterisk-users] no ringing tone making attended transfer whith an IAX client   Antonio Almodóvar
• [asterisk-users] Delay in posting of messages to list   David Boyd
• [asterisk-users] FX Dialing Odd   Eric "ManxPower" Wieling
• [asterisk-users] FX Dialing Odd   Eric "ManxPower" Wieling
• [asterisk-users] Calls being dropped   Compnet Bobby
• [asterisk-users] cisco 7940 and auto-answer (aastra 480i vs 7940)   Julian Lyndon-Smith
• [asterisk-users] realtime ldap peer matching   Caio Zanolla
• [asterisk-users] Asterisk 1.4.4 Segfaults with asterisk-ooh323 from addons-1.4.1   Bruce Ferrell
• [asterisk-users] Delay in posting of messages to list   John Novack
• [asterisk-users] Delay in posting of messages to list   Erik Anderson
• [asterisk-users] answer a voip call, play info.   Matthew Pease
• [asterisk-users] Detecting card on the PCI Slot   Sanjay Rajdev
• [asterisk-users] cisco 7940 and auto-answer (aastra 480i vs 7940)   Justin Moore
• [asterisk-users] Get calling channel before pickup   Marcus Carlson
• [asterisk-users] Help with IAX   Malcom Kemp
• [asterisk-users] Detecting card on the PCI Slot   Brandon Kruse
• [asterisk-users] Detecting card on the PCI Slot   Gordon Henderson
• [asterisk-users] Detecting card on the PCI Slot   Tzafrir Cohen
• [asterisk-users] Get calling channel before pickup   Eric "ManxPower" Wieling
• [asterisk-users] cisco 7940 and auto-answer (aastra 480i vs 7940)   Dave Bour
• [asterisk-users] yum om centos   Rob Townley
• [asterisk-users] Detecting card on the PCI Slot   Sanjay Rajdev
• [asterisk-users] FX Dialing Odd   Rob Schall
• [asterisk-users] Detecting card on the PCI Slot   Sanjay Rajdev
• [asterisk-users] debug logs   ram
• [asterisk-users] realtime ldap peer matching   Gavin Henry
• [asterisk-users] debug logs   Mike Lynchfield
• [asterisk-users] Calls being dropped   Mike Lynchfield
• [asterisk-users] Auto Dial Problem   aslay-pinwee
• [asterisk-users] Oddity   Mike Hammett
• [asterisk-users] addqueuemember recording and reporting problems   Jared Smith
• [asterisk-users] Debug meetme   Christopher Aloi
• [asterisk-users] Aastra 57i / 57i CT phones fail to re-register   Julian Yap
• [asterisk-users] Digium Card   Stephen Bosch
• [asterisk-users] Noise on FXS ports (Sangoma)   Stephen Bosch
• [asterisk-users] Noise on FXS ports (Sangoma)   Stephen Bosch
• [asterisk-users] chan_sip.c: That's odd... Got a response on a call we dont know about.   emdeex
• [asterisk-users] Noise on FXS ports (Sangoma)   Stephen Bosch
• [asterisk-users] debug logs   Tzafrir Cohen
• [asterisk-users] wifi sip phone real-world experiences?   F6HQZ
• [asterisk-users] spa 3102 configuration   damiano bertuna
• [asterisk-users] cepstral TTS and app_swift   Julian Lyndon-Smith
• [asterisk-users] wifi sip phone real-world experiences?   Paul Hayes
• [asterisk-users] Debug meetme   Adrian Marsh
• [asterisk-users] background dialing   Thomas Stein
• [asterisk-users] background dialing   Rizwan Hisham
• [asterisk-users] background dialing   Rizwan Hisham
• [asterisk-users] IS_REGISTERED from dialplan   dima
• [asterisk-users] Calls being dropped   Andreas Brodmann
• [asterisk-users] background dialing   Thomas Stein
• [asterisk-users] IAX2 Trunk No Sound   Arun Kumar
• [asterisk-users] Problem to park the call with #700   lavarini at sci.univr.it
• [asterisk-users] Meetme define context   Chris Blunt
• [asterisk-users] background dialing   Rizwan Hisham
• [asterisk-users] Dynamically adding Context in dialplan?   Nasir Iqbal
• [asterisk-users] background dialing   Nasir Iqbal
• [asterisk-users] realtime ldap peer matching   Caio Zanolla
• [asterisk-users] Re: Meetme define context   Tony Mountifield
• [asterisk-users] wifi sip phone real-world experiences?   Andrew Kohlsmith
• [asterisk-users] spa 3102 configuration   Mandeep Singh Bhabha
• [asterisk-users] X100P Clone   Ronaldo
• [asterisk-users] X100P Clone   Henry Cobb
• [asterisk-users] X100P Clone   Tim Litwiller
• [asterisk-users] NAT   Iban Lopetegi Zinkunegi
• [asterisk-users] Strange beeping sound during fax initiate session   Philipp Ott
• [asterisk-users] NAT   Henry Cobb
• [asterisk-users] Asterisk on x64   Akpome Akpoguma
• [asterisk-users] addqueuemember recording and reporting   Jordan Novak
• [asterisk-users] Asterisk on x64   --
• [asterisk-users] NAT   Cosmin Prund
• [asterisk-users] spa 3102 incoming call   damiano bertuna
• [asterisk-users] AEL2 Includes in Macro...   Steve Murphy
• [asterisk-users] Calls being dropped   Rizwan Hisham
• [asterisk-users] spa 3102 incoming call   Leonardo Kamache (Gmail)
• [asterisk-users] addqueuemember recording and reporting   Andrey Solovjov
• [asterisk-users] Debug meetme   Christopher Aloi
• [asterisk-users] Chan_mobile issue   Jason Parker
• [asterisk-users] Digium Card   Noah Miller
• [asterisk-users] Noise on FXS ports (Sangoma)   shadowym
• [asterisk-users] Digium Card   David Gomillion
• [asterisk-users] answer a voip call, play info.   Mojo with Horan & Company, LLC
• [asterisk-users] answer a voip call, play info.   Mojo with Horan & Company, LLC
• [asterisk-users] IAX2 Trunk No Sound   Noah Miller
• [asterisk-users] Get calling channel before pickup   Mojo with Horan & Company, LLC
• [asterisk-users] NAT   Gordon Henderson
• [asterisk-users] NAT   Tom Rymes
• [asterisk-users] IAX2 Trunk No Sound   Arun Kumar
• [asterisk-users] Cisco 7961G + 7914 Expansion Module   Eric Lubow
• [asterisk-users] Cisco 7961G + 7914 Expansion Module   Jason Parker
• [asterisk-users] answer a voip call, play info.   Matthew Pease
• [asterisk-users] cepstral TTS and app_swift   Mojo with Horan & Company, LLC
• [asterisk-users] Debug meetme   Adrian Marsh
• [asterisk-users] Hardware spec comparison   Adrian Marsh
• [asterisk-users] Training/Teaching our employees how to use Asterisk and phones   Martin Smith
• [asterisk-users] Where to find Polycom firmware with 330/320 support?   Mike
• [asterisk-users] Where to find Polycom firmware with 330/320support?   Martin Smith
• [asterisk-users] Changing the From field in Asterisk email/voicemail   Jim Suber
• [asterisk-users] zaptel on CENTOS servercd   Jim Suber
• [asterisk-users] Hardware spec comparison   Bobby Crawford
• [asterisk-users] Where to find Polycom firmware with 330/320 support?   Fuermann, Jason Bryce
• [asterisk-users] Training/Teaching our employees how to useAsterisk and phones   Jonathan Barratt
• [asterisk-users] Open CallerID Database? (late reply)   John Todd
• [asterisk-users] X100P Clone   Jared Smith
• [asterisk-users] Changing the From field in Asterisk email/voicemail   Jared Smith
• [asterisk-users] Hardware spec comparison   Gordon Henderson
• [asterisk-users] Changing the From field in Asterisk email/voicemail   Gordon Henderson
• [asterisk-users] Training/Teaching our employees how to useAsterisk and phones   Compnet Bobby
• [asterisk-users] Cisco 7961G + 7914 Expansion Module   Eric Lubow
• [asterisk-users] Training/Teaching our employees how to useAsterisk and phones   Eric Lubow
• [asterisk-users] Hardware spec comparison   Adrian Marsh
• [asterisk-users] Noise on FXS ports (Sangoma)   Stephen Bosch
• [asterisk-users] Training/Teaching our employees howto useAsterisk and phones   Jonathan Barratt
• [asterisk-users] Get calling channel before pickup   Marcus Carlson
• [asterisk-users] Set caller ID based on SIP source.   Alex Balashov
• [asterisk-users] Verizon Interconnection   Matt
• [asterisk-users] g729   Ed Nuñez
• [asterisk-users] Oddity   Mike Hammett
• [asterisk-users] Where to find Polycom firmware with 330/320 support?   Stephen Bosch
• [asterisk-users] Set caller ID based on SIP source.   Alexander Lopez
• [asterisk-users] Set caller ID based on SIP source.   Alex Balashov
• [asterisk-users] Noise on FXS ports (Sangoma)   Jorge Mendoza
• [asterisk-users] Recomender Server specs for 250 con-current calls   Vidura Senadeera
• [asterisk-users] Set caller ID based on SIP source.   Brian McManus
• [asterisk-users] Outlook dialing   Dean Collins
• [asterisk-users] Noise on FXS ports (Sangoma)   shadowym
• [asterisk-users] Outlook dialing   Dovid B
• [asterisk-users] Problem to park the call with #700   Dovid B
• [asterisk-users] Noise on FXS ports (Sangoma)   Stephen Bosch
• [asterisk-users] background dialing   Dovid B
• [asterisk-users] Noise on FXS ports (Sangoma)   Stephen Bosch
• [asterisk-users] Dynamically adding Context in dialplan?   Dovid B
• [asterisk-users] Outlook dialing   Paul Hales
• [asterisk-users] background dialing   Thomas Stein
• [asterisk-users] Queue problem   Elmar Haneke
• [asterisk-users] Zaptel and Libpri Compilation   bilal ghayyad
• [asterisk-users] Re: Problem to park the call with #700   lavarini at sci.univr.it
• [asterisk-users] Noise on FXS ports (Sangoma)   François Delawarde
• [asterisk-users] g729   Jaswinder Singh
• [asterisk-users] Asterisk call quality detection   carl Lougher
• [asterisk-users] Where to find Polycom firmware with 330/320 support?   Mandeep Singh Bhabha
• [asterisk-users] WiFi SIP phones   Alberto Sagredo (M)
• [asterisk-users] CDR changes in 1.4.3?   Atis
• [asterisk-users] WiFi SIP phones   Alberto Pastore
• [asterisk-users] Asterisk call quality detection   Henry Cobb
• [asterisk-users] X100P Clone   Henry Cobb
• [asterisk-users] Re: Verizon Interconnection   Matt
• [asterisk-users] Strange beeping sound during fax initiate session   Philipp Ott
• [asterisk-users] shorting flash time   Steve Kennedy
• [asterisk-users] g729   Ed Nunez
• [asterisk-users] Duplicate UNIQUEID on CDR   Steve Davies
• [asterisk-users] Astmanproxy   Steve Davies
• [asterisk-users] Asterisk call quality detection   carl
• [asterisk-users] Multiple Ethernet Interfaces   Andre Wangler
• [asterisk-users] WiFi SIP phones   olivier.taylor
• [asterisk-users] blades?   Dean Collins
• [asterisk-users] Asterisk call quality detection   Matt
• [asterisk-users] Re: Verizon Interconnection   Klaverstyn, David C
• [asterisk-users] Re: Verizon Interconnection   laurent schweizer
• [asterisk-users] g729   Ed Nuñez
• [asterisk-users] Where to find Polycom firmware with 330/320 support?   Fuermann, Jason Bryce
• [asterisk-users] blades?   dubrowin.44628913 at bloglines.com
• [asterisk-users] PRI Partial Re-Rounting   Jon Schøpzinsky
• [asterisk-users] blades?   --
• [asterisk-users] Voip-info.org   Ed Nuñez
• [asterisk-users] shorting flash time   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk call quality detection   carl
• [asterisk-users] Asterisk call quality detection   Henry Cobb
• [asterisk-users] blades?   Jon Pounder
• [asterisk-users] Noise on FXS ports (Sangoma)   John Novack
• [asterisk-users] Re: Verizon Interconnection   carl
• [asterisk-users] Re: Verizon Interconnection   Gunnar Schaller
• [asterisk-users] Noise on FXS ports (Sangoma)   Jorge Mendoza
• [asterisk-users] asterisk 1.2.18 problems...   Angel Luis Martinez
• [asterisk-users] Dial plan inquiry using GotoIf()   Steve Murphy
• [asterisk-users] Re: Verizon Interconnection   Matt
• [asterisk-users] Outlook dialing   Martin Smith
• [asterisk-users] PRI Partial Re-Rounting   Anthony Francis
• [asterisk-users] Voip-info.org   Roger Schreiter
• [asterisk-users] PRI Partial Re-Rounting   Eric "ManxPower" Wieling
• [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *   Steve Murphy
• [asterisk-users] Where to find Polycom firmware with 330/320 support?   Jessee J Holmes
• [asterisk-users] Voip-info.org   Leonardo Kamache (Gmail)
• [asterisk-users] shorting flash time   Patrick
• [asterisk-users] Voip-info.org   Justin Moore
• [asterisk-users] shorting flash time   Steve Kennedy
• [asterisk-users] Re: Verizon Interconnection   James Coberly
• [asterisk-users] Polycom 320 messages   Mike Hammett
• [asterisk-users] Re: Verizon Interconnection   Mike Hammett
• [asterisk-users] g729   Jaswinder Singh
• [asterisk-users] Voip-info.org   Jaswinder Singh
• [asterisk-users] g729   Jaswinder Singh
• [asterisk-users] Voip-info.org   James H Thompson
• [asterisk-users] Voip-info.org   Paul
• [asterisk-users] Voip-info.org   Dave Bour
• [asterisk-users] Voip-info.org   Luis Morales
• [asterisk-users] TCP<->UDP SIP proxy?   Yehavi Bourvine +972-8-9489444
• [asterisk-users] any codec passthru mode   Jaswinder Singh
• [asterisk-users] Voip-info.org   Atis
• [asterisk-users] blades?   Mike Lynchfield
• [asterisk-users] Outlook dialing   Mike Lynchfield
• [asterisk-users] shorting flash time   Mojo with Horan & Company, LLC
• [asterisk-users] Dial plan inquiry using GotoIf()   Mike Lynchfield
• [asterisk-users] Voip-info.org   Mike Lynchfield
• [asterisk-users] Voip-info.org   Mojo with Horan & Company, LLC
• [asterisk-users] Asterisk call quality detection   Mike Lynchfield
• [asterisk-users] Noise on FXS ports (Sangoma)   shadowym
• [asterisk-users] Needed changes in Asterisk to change the SIP port to 5062   Crazy Boy
• [asterisk-users] Voip-info.org   Guillermo Salas M.
• [asterisk-users] Re: Verizon Interconnection   laurent schweizer
• [asterisk-users] zaptel make problem   Malcom Kemp
• [asterisk-users] Asterisk call quality detection   Willy Wouters
• [asterisk-users] Re: Verizon Interconnection   Matt
• [asterisk-users] cepstral TTS and app_swift   Mike Lynchfield
• [asterisk-users] Voip-info.org   Compnet Bobby
• [asterisk-users] PRI Partial Re-Rounting   John Treble
• [asterisk-users] Queue Job   Jason Adams
• [asterisk-users] meetme realtime   ram
• [asterisk-users] TCP<->UDP SIP proxy?   Victor Toofic
• [asterisk-users] TCP<->UDP SIP proxy?   Alex Robar
• [asterisk-users] Verizon Interconnection   Jared Smith
• [asterisk-users] TCP<->UDP SIP proxy?   Remco Post
• [asterisk-users] TCP<->UDP SIP proxy?   Kristian Kielhofner
• [asterisk-users] blades?   Jon Pounder
• [asterisk-users] Record CDR in a Oracle database   Everton Goularth
• [asterisk-users] TCP<->UDP SIP proxy?   Jaswinder Singh
• [asterisk-users] Asterisk call quality detection   Jared Smith
• [asterisk-users] Scaling Asterisk: High volume benchmarks (0 to 450 calls)   Matthew J. Roth
• [asterisk-users] Stanaphone/Asterisk issue: No Audio with SIP to only one provider when switching servers   Alejandro Lengua
• [asterisk-users] Noise on FXS ports (Sangoma)   John Novack
• [asterisk-users] Voicemail marking messages as Old   Adrian A
• [asterisk-users] Needed changes in Asterisk to change the SIP port to 5062   Philipp Kempgen
• [asterisk-users] PRI Partial Re-Rounting   Matthew Fredrickson
• [asterisk-users] Needed changes in Asterisk to change the SIP port to 5062   ~Russell
• [asterisk-users] Re: Verizon Interconnection   laurent schweizer
• [asterisk-users] zaptel make problem   Tzafrir Cohen
• [asterisk-users] Voip-info.org   Jaswinder Singh
• [asterisk-users] Phantom calls: Detecting hangup quicker   Gavin Henry
• [asterisk-users] Needed changes in Asterisk to change the SIP port to 5062   Jaswinder Singh
• [asterisk-users] cepstral TTS and app_swift   Julian Lyndon-Smith
• [asterisk-users] Voip-info.org   Remco Post
• [asterisk-users] Polycoms lose registration and won't re-register   ewr at erols.com
• [asterisk-users] PRI Partial Re-Rounting   Anthony Francis
• [asterisk-users] X100P Clone   John Novack
• [asterisk-users] Console duplicate output problem   Barton Fisher
• [asterisk-users] Reload in 1.4 clears regexten   Douglas Garstang
• [asterisk-users] Voicemail marking messages as Old   Jason Parker
• [asterisk-users] Sending multiline SMS   Patrick Zwahlen
• [asterisk-users] Slow list   Philipp Kempgen
• [asterisk-users] Stanaphone/Asterisk issue: No Audio with SIP to only one provider when switching servers   Hadar Pedhazur
• [asterisk-users] SetAccount in extensions.conf   Matt
• [asterisk-users] 1.4 Zaptel/Sangoma Issues on CentOS   Steve Totaro
• [asterisk-users] meetme realtime   Carlos Chavez
• [asterisk-users] Asterisk call quality detection   Andrew Kohlsmith
• [asterisk-users] 1.4 Zaptel/Sangoma Issues on CentOS   Bruce Reeves
• [asterisk-users] 1.4 Zaptel/Sangoma Issues on CentOS   Bruce Reeves
• [asterisk-users] Queue problem   BJ Weschke
• Solved: [asterisk-users] Subscriptions to Agent hints stopped working after a changing the numbering   Alexander Topolanek
• [asterisk-users] Best Codec   Davis Sylvester III
• [asterisk-users] Voicemail marking messages as Old   John Novack
• "Me Too" mails DOSsing the list (was: Re: [asterisk-users] Voip-info.org)   Brian Capouch
• [asterisk-users] Noise on FXS ports (Sangoma)   Stephen Bosch
• [asterisk-users] Solved: [SetAccount in extensions.conf]   Matt
• [asterisk-users] Noise on FXS ports (Sangoma)   Stephen Bosch
• [asterisk-users] Phantom calls: Detecting hangup quicker   Stephen Bosch
• [asterisk-users] meetme realtime   ram
• [asterisk-users] SIP buddy watch   clive.chan(Alpha Trilogies Networks)
• [asterisk-users] Scaling Asterisk: High volume benchmarks (0 to 450 calls)   Remco Post
• [asterisk-users] Slow list   Remco Post
• [asterisk-users] iax trunking on OpenBSD   Sebastian Reitenbach
• [asterisk-users] voice activated recording   Vieri
• [asterisk-users] Linksys WRTP54G-NA with SIP   Thomas Kenyon
• [asterisk-users] iax trunking on OpenBSD   Arun Kumar
• [asterisk-users] call Hold event asterisk   sathish s
• [asterisk-users] Phantom calls: Detecting hangup quicker   Gavin Henry
• [asterisk-users] PRI Partial Re-Rounting   Jon Schøpzinsky
• [asterisk-users] X100P Clone   Henry Cobb
• [asterisk-users] Dial plan inquiry using GotoIf()   Gordon Henderson
• [asterisk-users] Sending multiline SMS   Anselm Martin Hoffmeister
• [asterisk-users] Best Codec   Gordon Henderson
• [asterisk-users] MP3 as voicemail format   Rizwan Hisham
• [asterisk-users] AddQueueMember vs AgentCallbackLogin   Atis
• [asterisk-users] AddQueueMember vs AgentCallbackLogin   Julian Lyndon-Smith
• [asterisk-users] X100P Clone   Tzafrir Cohen
• [asterisk-users] AddQueueMember vs AgentCallbackLogin   Atis
• [asterisk-users] Reload in 1.4 clears regexten   Watkins, Bradley
• [asterisk-users] atxfer not working   gincantalupo
• [asterisk-users] Zaptel and Libpri compilation   bilal ghayyad
• [asterisk-users] Zaptel and Libpri compilation   Tzafrir Cohen
• [asterisk-users] Realtime Agents.conf   srinivas Antarvedi
• [asterisk-users] Need help on Text entry for asterisk through touch pad   rajesh koniki
• [asterisk-users] Need help on Text entry for asterisk through touchpad   rajesh koniki
• [asterisk-users] Sending multiline SMS   Patrick Zwahlen
• [asterisk-users] Meet Me video conferencing   Khaled Chehab
• [asterisk-users] Need help on Text entry applicaon   rajesh koniki
• [asterisk-users] zaptel make problem   Malcom Kemp
• [asterisk-users] Polycom phone registration problem   Laurent CARON
• [asterisk-users] Polycom phone registration problem   Rob Schall
• [asterisk-users] zaptel make problem   Tzafrir Cohen
• [asterisk-users] Wireless IP Phone with external Telephone Book   Drew Gibson
• [asterisk-users] Polycom phone registration problem   Laurent CARON
• [asterisk-users] zaptel make problem   Malcom Kemp
• [asterisk-users] PRI Partial Re-Rounting   John Treble
• [asterisk-users] PRI Partial Re-Rounting   John Treble
• [asterisk-users] Noise on FXS ports (Sangoma)   John Novack
• [asterisk-users] Polycom phone registration problem   Laurent CARON
• [asterisk-users] call Hold event asterisk   Lee Jenkins
• [asterisk-users] Reload in 1.4 clears regexten   Douglas Garstang
• [asterisk-users] 1.4 Zaptel/Sangoma Issues on CentOS   Jared Smith
• [asterisk-users] DUNDi and reinvites...   Douglas Garstang
• [asterisk-users] meetme realtime   Carlos Chavez
• [asterisk-users] meetme realtime   ram
• [asterisk-users] Polycom phone registration problem   Rob Schall
• [asterisk-users] g729   Ed Nuñez
• [asterisk-users] PRI Partial Re-Rounting   Matthew Fredrickson
• [asterisk-users] PRI Partial Re-Rounting   Watkins, Bradley
• [asterisk-users] Reload in 1.4 clears regexten   Watkins, Bradley
• [asterisk-users] Bridged PRI calls - processor involvement?   Steve Hanselman
• [asterisk-users] Noise on FXS ports (Sangoma)   Stephen Bosch
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Douglas Garstang
• [asterisk-users] Noise on FXS ports (Sangoma)   John Novack
• [asterisk-users] meetme realtime   Carlos Chavez
• [asterisk-users] Polycoms lose registration and won't re-register   ewr at erols.com
• [asterisk-users] IAX2 Trunk No Sound   Tim Panton
• [asterisk-users] Bridged PRI calls - processor involvement?   Matthew Fredrickson
• [asterisk-users] DUNDi and reinvites...   Jared Smith
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Doug
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Douglas Garstang
• [asterisk-users] Hardware spec comparison   Tim Panton
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Steve Edwards
• [asterisk-users] Chanspy severe sound problems   Jesus Mogollon
• [asterisk-users] g729   Ed Nuñez
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Douglas Garstang
• [asterisk-users] DUNDi and reinvites...   Douglas Garstang
• [asterisk-users] meetme realtime   ram
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Jon Pounder
• [asterisk-users] DUNDi and reinvites...   Douglas Garstang
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Jared Smith
• [asterisk-users] custom cdr fields and cdr_mysql, howto?   JR Richardson
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Steve Edwards
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Steve Edwards
• [asterisk-users] getting at ${CALLERIDNUM}   Matthew Pease
• [asterisk-users] Scaling Asterisk: High volume benchmarks (0 to 450 calls)   Matthew J. Roth
• [asterisk-users] Noise on FXS ports (Sangoma)   Tzafrir Cohen
• [asterisk-users] Best Codec   Ricardo Martins
• [asterisk-users] Re: Meet Me video conferencing   Steven
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Jared Smith
• [asterisk-users] getting at ${CALLERIDNUM}   Jared Smith
• [asterisk-users] getting at ${CALLERIDNUM}   John covici
• [asterisk-users] Q931 Error with H323   Dovid B
• [asterisk-users] DUNDi and reinvites...   Eric "ManxPower" Wieling
• [asterisk-users] Noise on FXS ports (Sangoma)   John Novack
• [asterisk-users] getting at ${CALLERIDNUM}   Eric "ManxPower" Wieling
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Steve Edwards
• [asterisk-users] getting at ${CALLERIDNUM}   Eric "ManxPower" Wieling
• [asterisk-users] agi with java?   Matthew Pease
• [asterisk-users] 3 questions - variables, upgrading, and IRC   Nick Seraphin
• [asterisk-users] getting at ${CALLERIDNUM}   Matthew Pease
• [asterisk-users] DUNDi and reinvites...   Douglas Garstang
• [asterisk-users] RFC-3389 problem   M vidyasagar
• [asterisk-users] agi with java?   Stefan Reuter
• [asterisk-users] Noise on FXS ports (Sangoma)   Stephen Bosch
• [asterisk-users] IAX-configuration   Andre Wangler
• [asterisk-users] DUNDi and reinvites...   Jared Smith
• [asterisk-users] 3 questions - variables, upgrading, and IRC   Jared Smith
• [asterisk-users] getting at ${CALLERIDNUM}   Jared Smith
• [asterisk-users] Duplicate UNIQUEID on CDR   Steve Murphy
• [asterisk-users] IAX trunk with dynamic IPs   Ronaldo Z. Afonso
• [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)   Nick Seraphin
• [asterisk-users] CDR changes in 1.4.3?   Steve Murphy
• [asterisk-users] RFC-3389 problem   Eric "ManxPower" Wieling
• [asterisk-users] Best Codec   Henry Cobb
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Mattt
• [asterisk-users] Console duplicate output problem   Barton Fisher
• [asterisk-users] getting at ${CALLERIDNUM}   Matthew Pease
• [asterisk-users] getting at ${CALLERIDNUM}   Matthew Pease
• [asterisk-users] Semi OT - Cable products suppliers   groups at soundscapestudio.net
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Nick Seraphin
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Stewart Nelson
• [asterisk-users] Semi OT - Cable products suppliers   Nick Seraphin
• [asterisk-users] 3 questions - variables, upgrading, and IRC   Nick Seraphin
• [asterisk-users] Semi OT - Cable products suppliers   Al Bochter
• [asterisk-users] iax trunking on OpenBSD   Sebastian Reitenbach
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Steve Edwards
• [asterisk-users] Provisioning Linksys PAP2T ATA's   Mattt
• [asterisk-users] Bridged PRI calls - processor involvement?   Steve Hanselman
• [asterisk-users] Asterisk & MS RTC Library & Ethernet Capacity   Asterisk
• [asterisk-users] Unexpected behaviour shown by "meetme kick confno usernumber"   Muhammad Raza
• [asterisk-users] Asterisk & MS RTC Library & Ethernet Capacity   Gordon Henderson
• [asterisk-users] Noise on FXS ports (Sangoma)   Tzafrir Cohen
• [asterisk-users] Asterisk & MS RTC Library & Ethernet Capacity   Asterisk
• [asterisk-users] Polycom phone registration problem   Laurent CARON
• [asterisk-users] tdd.c - How does one use this code?   Don Fanning
• [asterisk-users] DUNDi and reinvites...   Bryan Laird
• [asterisk-users] Noise on FXS ports (Sangoma)   John Novack
• [asterisk-users] Asterisk, NAT and canreinvite=yes   lokotes2
• [asterisk-users] agi with java?   Gary Chen
• [asterisk-users] agi with java?   Lenz
• [asterisk-users] Noise on FXS ports (Sangoma)   Stephen Bosch
• [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)   Jared Smith
• [asterisk-users] getting at ${CALLERIDNUM}   Jared Smith
• [asterisk-users] call problem...   Carlos Jerónimo
• [asterisk-users] Changing the messages that are played when a user is unavailable/busy   Timothy Parez
• [asterisk-users] call problem...   Tzafrir Cohen
• [asterisk-users] Bridged PRI calls - processor involvement?   Matthew Fredrickson
• [asterisk-users] Not getting CID Name from PRI   Kyle Sexton
• [asterisk-users] Q931 Error with H323   John Treble
• [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)   Steve Edwards
• [asterisk-users] Hot GXP-2000   Carlos Chavez
• [asterisk-users] Changing the messages that are played when a user is unavailable/busy   Eric "ManxPower" Wieling
• [asterisk-users] Best Codec   Chris Bagnall
• [asterisk-users] No/unknown event '0' on timer   Doug Lytle
• [asterisk-users] Console duplicate output problem   Eric "ManxPower" Wieling
• [asterisk-users] Best Codec   Ricardo Martins
• [asterisk-users] agi with java?   Lee Jenkins
• [asterisk-users] Hot GXP-2000   Jessee J Holmes
• [asterisk-users] choppy sound with playback, background, etc... but not with musiconhold   Paco Brufal
• [asterisk-users] Bridged PRI calls - processor involvement?   Steve Hanselman
• [asterisk-users] Best Codec   Luki
• [asterisk-users] Re: Not getting CID Name from PRI   Kyle Sexton
• [asterisk-users] Write to multiple databases as redundancy scheme   rjcarvalho at reit.up.pt
• [asterisk-users] Changing the messages that are played when a user is unavailable/busy   Justin Moore
• [asterisk-users] call Hold event asterisk   Anthony Francis
• [asterisk-users] Write to multiple databases as redundancy scheme   Chris Mason (Lists)
• [asterisk-users] Write to multiple databases as redundancy scheme   Bobby Crawford
• [asterisk-users] Write to multiple databases as redundancy scheme   Bryan Laird
• [asterisk-users] Write to multiple databases as redundancy scheme   Doug Lytle
• [asterisk-users] Write to multiple databases as redundancy scheme   Justin Moore
• [asterisk-users] Console duplicate output problem   Barton Fisher
• [asterisk-users] agi with java?   Chris Bagnall
• [asterisk-users] Write to multiple databases as redundancy scheme   Anthony Francis
• [asterisk-users] getting at ${CALLERIDNUM}   Phil Reynolds
• [asterisk-users] Bridged PRI calls - processor involvement?   Matthew Fredrickson
• [asterisk-users] Changing the messages that are played when a user is unavailable/busy   Philipp Kempgen
• [asterisk-users] call Hold event asterisk   Steve Murphy
• [asterisk-users] Can Asterisk RAS?   Christopher Dobbs
• [asterisk-users] Write to multiple databases as redundancy scheme   Watkins, Bradley
• [asterisk-users] Log interpretation   Adrian Marsh
• [asterisk-users] Re: custom cdr fields and cdr_mysql, howto?   JR Richardson
• [asterisk-users] Re: Write to multiple databases as redundancy scheme   JR Richardson
• [asterisk-users] Write to multiple databases as redundancy scheme   David Gomillion
• [asterisk-users] call problem...   Carlos Jerónimo
• [asterisk-users] Replacing SX-2000 Centigram Voicemail with Asterisk?   George Pajari
• [asterisk-users] Can Asterisk RAS?   Jared Smith
• [asterisk-users] Can Asterisk RAS?   Michelle Dupuis
• [asterisk-users] call problem...   Tzafrir Cohen
• [asterisk-users] choppy sound with playback, background, etc... but not with musiconhold   Gordon Henderson
• [asterisk-users] Zaptel 1.2.18 and 1.4.3 released!   The Asterisk Development Team
• [asterisk-users] choppy sound with playback, background, etc... but not with musiconhold   Paco Brufal
• [asterisk-users] Bad Echo between SIP calls   Deepak Naidu
• [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING   Alvaro Parres
• [asterisk-users] Bad Echo between SIP calls   Alex Balashov
• [asterisk-users] Asterisk 1.4 with Unicall   Carlos Chavez
• [asterisk-users] FW: Delivery Status Notification(Failure)   Steve Totaro
• [asterisk-users] Changing the messages that are played when a user is unavailable/busy   Timothy Parez
• [asterisk-users] SIP Transit problem   Gary Mensenares
• [asterisk-users] Bad Echo between SIP calls   Deepak Naidu
• [asterisk-users] CDR accuracy   clive.chan(Alpha Trilogies Networks)
• [asterisk-users] Softphone for smartphone such as Nokia N90 / 93 / N95   Asterisk guy
• [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)   Nick Seraphin
• [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)   Nick Seraphin
• [asterisk-users] Asterisk 1.4 with Unicall   Moises Silva
• [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING   Moises Silva
• [asterisk-users] Bad Echo between SIP calls   Stephen Davies
• [asterisk-users] Asterisk Users Conference Friday: New Asterisk Book and a visit from JerJer of Nufone   randulo
• [asterisk-users] Asterisk & MS RTC Library & Ethernet Capacity   Stephen Davies
• [asterisk-users] Bad Echo between SIP calls   Deepak Naidu
• [asterisk-users] Bad Echo between SIP calls   Steve Totaro
• [asterisk-users] Bad Echo between SIP calls   Steve Totaro
• [asterisk-users] Bridged PRI calls - processor involvement?   Steve Totaro
• [asterisk-users] Is There any Asterisk TODO(Developer side) list Available   Ibrar Ahmed
• [asterisk-users] No sound, problem is not a NAT   Carlos Jerónimo
• [asterisk-users] R2 Argentina   Oscar Carriles
• [asterisk-users] Linksys 941/942 reboot and persistent MWI   Bruce Komito
• [asterisk-users] IAX2 Trunk No Sound   Noah Miller
• [asterisk-users] Bad Echo between SIP calls   Eric "ManxPower" Wieling
• [asterisk-users] IAX trunk with dynamic IPs   Noah Miller
• [asterisk-users] No sound, problem is not a NAT   Ricardo Martins
• [asterisk-users] No sound, problem is not a NAT   Noah Miller
• [asterisk-users] Re: SIP & NAT ...   Gordon Henderson
• [asterisk-users] Bad Echo between SIP calls   Deepak Naidu
• [asterisk-users] Bad Echo between SIP calls   Deepak Naidu
• [asterisk-users] Bad Echo between SIP calls   Steve Underwood
• [asterisk-users] Hot GXP-2000   Dovid B
• [asterisk-users] H.323 trunk between MD110 and Asterisk   asterisk-users at rogg.is
• [asterisk-users] How to tell what codec is used for each end of a call MD110->H323->SIP   asterisk-users at rogg.is
• [asterisk-users] remove   Julio lopez
• [asterisk-users] IAX trunk with dynamic IPs   Ronaldo Z. Afonso
• [asterisk-users] IAX trunk with dynamic IPs   Justin Moore
• [asterisk-users] IAX trunk with dynamic IPs   Matt
• [asterisk-users] OT: CallManager ANI restamp.   Alex Balashov
• [asterisk-users] OT: CallManager ANI restamp.   Mattt
• [asterisk-users] IAX trunk with dynamic IPs   Ronaldo Z. Afonso
• [asterisk-users] OT: CallManager ANI restamp.   Alex Balashov
• [asterisk-users] Polycom 301 vs. 330   Lee Jenkins
• [asterisk-users] Polycom 301 vs. 330   Jeff Davis
• [asterisk-users] Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event   Lee Jenkins
• [asterisk-users] Polycom 301 vs. 330   Lee Jenkins
• [asterisk-users] Bad Echo between SIP calls   C F
• [asterisk-users] ast_dynamic_str_thread_build_va() is defined with 6 args but only called with 5 args??   Frank Tarczynski
• [asterisk-users] Hot GXP-2000   Bill Hackensack
• [asterisk-users] Noise on FXS ports (Sangoma)   Stephen Bosch
• [asterisk-users] Bad Echo between SIP calls   Deepak Naidu
• [asterisk-users] Bad Echo between SIP calls   C F
• [asterisk-users] Hot GXP-2000   C F
• [asterisk-users] Bad Echo between SIP calls   Darryl Dunkin
• [asterisk-users] agi with java?   Lenz
• [asterisk-users] IAX trunk with dynamic IPs   Jaswinder Singh
• [asterisk-users] Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event   Jeff Davis
• [asterisk-users] IAX trunk with dynamic IPs   Ronaldo Z. Afonso
• [asterisk-users] IAX Peers show command   Ronaldo Z. Afonso
• [asterisk-users] agi with java?   Tim Panton
• [asterisk-users] Blocking 900 calls   Larry Alkoff
• [asterisk-users] Blocking 900 calls   Luki
• [asterisk-users] Blocking 900 calls   Richard Lyman
• [asterisk-users] Hot GXP-2000   shadowym
• [asterisk-users] IAX trunk with dynamic IPs   Jaswinder Singh
• [asterisk-users] ast_dynamic_str_thread_build_va() is defined with 6 args but only called with 5 args??   Moises Silva
• [asterisk-users] best format for audio via asterisk...   Matthew Pease
• [asterisk-users] Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event   Lee Jenkins
• [asterisk-users] call problem...   Christopher Dobbs
• [asterisk-users] RE: Asterisk RAS   Christopher Dobbs
• [asterisk-users] basic asterisk knowledge   Khaled Chehab
• [asterisk-users] Going to VON Stockholm? Meet you at the Asterisk BOF!   Olle E Johansson
• [asterisk-users] best format for audio via asterisk...   Gordon Henderson
• [asterisk-users] basic asterisk knowledge   Gordon Henderson
• [asterisk-users] Bridged PRI calls - processor involvement?   Steve Hanselman
• [asterisk-users] Bridged PRI calls - processor involvement?   Steve Hanselman
• [asterisk-users] Bridged PRI calls - processor involvement?   Gordon Henderson
• [asterisk-users] Bridged PRI calls - processor involvement?   Steve Hanselman
• [asterisk-users] No Audio with Gtalk   Charles Wang
• [asterisk-users] MOH Problems.   Klaverstyn, David C
• [asterisk-users] No Audio with Gtalk   Michael Zoller
• [asterisk-users] MOH Problems.   Thomas Stein
• [asterisk-users] CDR on transfers of called ZAP channel   Gunnar Schaller
• [asterisk-users] Searchable List Archives?   Matthew Rubenstein
• [asterisk-users] Help on text entry. using asterisk.   rajesh koniki
• [asterisk-users] (no subject)   rajesh koniki
• [asterisk-users] Searchable List Archives?   Per Jessen
• [asterisk-users] change moh during a call?   Thomas Stein
• [asterisk-users] Searchable List Archives?   Per Jessen
• [asterisk-users] Searchable List Archives?   Tzafrir Cohen
• [asterisk-users] Different ECs in Asterisk   Yonghua Fang
• [asterisk-users] How to tell what codec is used for each end of a call MD110->H323->SIP   mail-lists
• [asterisk-users] Asterisk as an SCCP client   Mark Phillips
• [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yieldinggains at high call volumes   Stelios Koroneos
• [asterisk-users] CDR on transfers of called ZAP channel   Steve Murphy
• [asterisk-users] IAX Peers show command   Anthony Francis
• [asterisk-users] IAX Peers show command   Ronaldo
• [asterisk-users] IAX Peers show command   Carlos Chavez
• [asterisk-users] IAX Peers show command   Vahan Yerkanian
• [asterisk-users] change moh during a call?   Anthony Francis
• [asterisk-users] IAX Peers show command   Joshua Colp
• [asterisk-users] Calls being dropped   Compnet Bobby
• [asterisk-users] CDR on transfers of called ZAP channel   David Boyd
• [asterisk-users] Console duplicate output problem   Mojo with Horan & Company, LLC
• [asterisk-users] Console duplicate output problem   Eric "ManxPower" Wieling
• [asterisk-users] Grandstream 4104 - Asterisk (Incoming Calls problem)   Antonopoulos Angelos
• [asterisk-users] Multiple ENUM entries and Asterisk fails to dial   rjcarvalho at reit.up.pt
• [asterisk-users] change moh during a call?   Eric "ManxPower" Wieling
• [asterisk-users] CallerID issues   Eric Lubow
• [asterisk-users] change moh during a call?   Atis
• [asterisk-users] Introduction to AGI programming   Kyle Sexton
• [asterisk-users] Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event   Lee Jenkins
• [asterisk-users] agi with java?   Lee Jenkins
• [asterisk-users] Introduction to AGI programming   Lee Jenkins
• [asterisk-users] AGI "RECORD FILE" for a video message   Jerry Geis
• [asterisk-users] Bad Echo between SIP calls   Matthew Fredrickson
• [asterisk-users] Multiple ENUM entries and Asterisk fails to dial   Remco Post
• [asterisk-users] sip show registry shows nothing   Nigel Kendrick
• [asterisk-users] Bad Echo between SIP calls   Deepak Naidu
• [asterisk-users] which Wifi SIP phones are the good ones   Deepak Naidu
• [asterisk-users] Bad Echo between SIP calls   Darryl Dunkin
• [asterisk-users] Bad Echo between SIP calls   Zeeshan Zakaria
• [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement   C F
• [asterisk-users] Bad Echo between SIP calls   Deepak Naidu
• [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement   Jon Pounder
• [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement   C F
• [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement   Alex Balashov
• [asterisk-users] Crashes with Spandsp, app_rxfax.c, and asterisk 1.4.4   Rob Ristroph
• [asterisk-users] change moh during a call?   Dovid B
• [asterisk-users] change moh during a call?   Thomas Stein
• [asterisk-users] which Wifi SIP phones are the good ones   Gordon Henderson
• [asterisk-users] change moh during a call?   Thomas Stein
• [asterisk-users] config files to mysql convertion   ram
• [asterisk-users] On multiple dial phones continue ringing after picked up   Yves Räber
• [asterisk-users] no dtmf pcom 650 only outbound calls   Dovid B
• [asterisk-users] Possible mysql database corruption   Nigel Kendrick
• [asterisk-users] SIP/NAT 1.2 1.4 questions   randulo
• [asterisk-users] SPA400 and asterisk   MBIT Technologies
• [asterisk-users] No audio after Dial with G option   Rosalinda Trevino Cadena
• [asterisk-users] SPA400 and asterisk   Alberto Sagredo (M)
• [asterisk-users] SPA400 and asterisk   MBIT Technologies
• [asterisk-users] which Wifi SIP phones are the good ones   Marco Mouta
• [asterisk-users] write some custom values to CDR table   rjcarvalho at reit.up.pt
• [asterisk-users] write some custom values to CDR table   Tomás Laureano Peralta Tormey
• [asterisk-users] write some custom values to CDR table   Doug Lytle
• [asterisk-users] call from ISDN   Josu Lazkano
• [asterisk-users] write some custom values to CDR table   rjcarvalho at reit.up.pt
• [asterisk-users] Warning on CLI   Josu Lazkano
• [asterisk-users] Changing the Caller ID   Shad Mortazavi
• [asterisk-users] Bad Echo between SIP calls   Matthew Fredrickson
• [asterisk-users] write some custom values to CDR table   Atis
• [asterisk-users] Bridge bug in 1.4?   Tony Plack
• [asterisk-users] Changing the Caller ID   Bobby Crawford
• [asterisk-users] Zombie SIP channels   Elmar Haneke
• [asterisk-users] [asterisk-tech] ChanSkype   Kyle Vorster
• [asterisk-users] AsterFax   Kyle Vorster
• [asterisk-users] Bridge bug in 1.4?   David Gomillion
• [asterisk-users] Bridge bug in 1.4?   rjcarvalho at reit.up.pt
• [asterisk-users] AsterFax   Jason Lixfeld
• [asterisk-users] Transfer caller direct to voicemail   Drew Gibson
• [asterisk-users] Transfer caller direct to voicemail   Leonardo Kamache (Gmail)
• [asterisk-users] Transfer caller direct to voicemail   Wes Baehr
• [asterisk-users] Changing the Caller ID   Anselm Martin Hoffmeister
• [asterisk-users] call from ISDN   Tzafrir Cohen
• [asterisk-users] HPEC and audioclipping   Olivier
• [asterisk-users] Changing the Caller ID   Eric "ManxPower" Wieling
• [asterisk-users] basic asterisk knowledge   Craig Guy
• [asterisk-users] call from ISDN   Josu Lazkano
• [asterisk-users] CDR on transfers of called ZAP channel   Gunnar Schaller
• [asterisk-users] Softphone behind NAT issues   Rob Schall
• [asterisk-users] choppy sound with playback, background, etc... but not with musiconhold   Matthew J. Roth
• [asterisk-users] Bad Echo between SIP calls   Mojo with Horan & Company, LLC
• [asterisk-users] Softphone behind NAT issues   Leonardo Kamache (Gmail)
• [asterisk-users] Asterisk Faxing   Kyle Vorster
• [asterisk-users] GotoIf Dialplan inquiry   Steve Finkelstein
• [asterisk-users] Softphone behind NAT issues   Tim Panton
• [asterisk-users] Softphone behind NAT issues   Rob Schall
• [asterisk-users] Softphone behind NAT issues   Rob Schall
• [asterisk-users] GotoIf Dialplan inquiry   C. Chad Wallace
• [asterisk-users] Bad Echo between SIP calls   Alex Crow
• [asterisk-users] call from ISDN   Tzafrir Cohen
• [asterisk-users] Answering machine detection after Dial()   Johannes Zweng
• [asterisk-users] GotoIf Dialplan inquiry   Gordon Henderson
• [asterisk-users] Softphone behind NAT issues   Rob Schall
• [asterisk-users] Record CDR in a Oracle database   Everton Goularth
• [asterisk-users] CDR changes in Trunk -- Transfers, CDRs, Life, and Everything   Steve Murphy
• [asterisk-users] GotoIf Dialplan inquiry   Mojo with Horan & Company, LLC
• [asterisk-users] Bridge bug in 1.4?   Tony Plack
• [asterisk-users] Re: [asterisk-dev] CDR changes in Trunk -- Transfers, CDRs, Life, and Everything   Stephen Davies
• [asterisk-users] GotoIf Dialplan inquiry   Doug Lytle
• [asterisk-users] Re: [asterisk-dev] CDR changes in Trunk -- Transfers, CDRs, Life, and Everything   David Boyd
• [asterisk-users] Gigabit SIP Phones   Olivier
• [asterisk-users] Realtime Meetme in 1.4   Tony Plack
• [asterisk-users] Gigabit SIP Phones   Erik Anderson
• [asterisk-users] Re: [asterisk-dev] CDR changes in Trunk -- Transfers, CDRs, Life, and Everything   Atis
• [asterisk-users] Gigabit SIP Phones   Andrew Latham
• [asterisk-users] Gigabit SIP Phones   Jon Pounder
• [asterisk-users] Gigabit SIP Phones   Paul
• [asterisk-users] anyway in meetme to mute all but one user?   John covici
• [asterisk-users] Realtime Meetme in 1.4   Dan Austin
• [asterisk-users] HPEC and audioclipping   Eric "ManxPower" Wieling
• [asterisk-users] Gigabit SIP Phones   Darrick Hartman
• [asterisk-users] Bad Echo between SIP calls   Matt
• [asterisk-users] Bad Echo between SIP calls   Matt
• [asterisk-users] Gigabit SIP Phones   Matthew Rubenstein
• [asterisk-users] Bad Echo between SIP calls   Tzafrir Cohen
• [asterisk-users] Bad Echo between SIP calls   Darryl Dunkin
• [asterisk-users] Bad Echo between SIP calls   Deepak Naidu
• [asterisk-users] Bad Echo between SIP calls   Darryl Dunkin
• [asterisk-users] Gigabit SIP Phones   James Harper
• [asterisk-users] Gigabit SIP Phones   Vamsi Pottangi
• [asterisk-users] Bad Echo between SIP calls   Eric "ManxPower" Wieling
• [asterisk-users] call from ISDN   Josu Lazkano
• [asterisk-users] zaphfc problem   Josu Lazkano
• [asterisk-users] SPA400 and asterisk   Nick Seraphin
• [asterisk-users] advanced asterisk logging   Harald Friessnegger
• [asterisk-users] Voicemail prob   Rizwan Hisham
• [asterisk-users] Voicemail prob   Gordon Henderson
• [asterisk-users] Bad Echo between SIP calls   Mindaugas Kuprys
• [asterisk-users] advanced asterisk logging   Atis
• [asterisk-users] Gigabit SIP Phones   Olivier
• [asterisk-users] HPEC and audioclipping   Olivier
• [asterisk-users] SPA400 and asterisk   Gergo Csibra
• [asterisk-users] re:zaphfc problem (Josu Lazkano)   Mauro Zanin
• [asterisk-users] problem starting asterisk, unable to load chan_zap   Sebastian Reitenbach
• [asterisk-users] re:zaphfc problem (Josu Lazkano)   Josu Lazkano
• [asterisk-users] Gigabit SIP Phones   Gordon Henderson
• [asterisk-users] problem starting asterisk, unable to load chan_zap   Doug Lytle
• [asterisk-users] Gigabit SIP Phones   Watkins, Bradley
• [asterisk-users] GotoIf Dialplan inquiry   Derek Whitten
• [asterisk-users] re:zaphfc problem (Josu Lazkano)   Tzafrir Cohen
• [asterisk-users] re:zaphfc problem (Josu Lazkano)   Tzafrir Cohen
• [asterisk-users] asterisk 1.2.18 problems...   Dovid B
• [asterisk-users] basic asterisk knowledge   Dovid B
• [asterisk-users] Using Modems with Asterisk   Lutgring, Sam
• [asterisk-users] Using Modems with Asterisk   Doug Lytle
• [asterisk-users] problem starting asterisk, unable to load chan_zap   Sebastian Reitenbach
• [asterisk-users] problem starting asterisk, unable to load chan_zap   Doug Lytle
• [asterisk-users] Polycom + Voicemail + Display message envelope in LCD   Martin Smith
• [asterisk-users] No audio after Dial with G option   Jaswinder Singh
• [asterisk-users] mISDN problem   Josu Lazkano
• [asterisk-users] problem starting asterisk, unable to load chan_zap   Sebastian Reitenbach
• [asterisk-users] HPEC and audioclipping   Matthew Fredrickson
• [asterisk-users] Using Modems with Asterisk   Jeremy Mann
• [asterisk-users] CallerID Presentation support for MFCR2   Oscar Carriles
• [asterisk-users] Gigabit SIP Phones   Matthew Rubenstein
• [asterisk-users] Using Modems with Asterisk   Jonathan Creasy
• [asterisk-users] Weird sip registration problem   Anthony Francis
• [asterisk-users] Polycom + Voicemail + Display message envelope inLCD   Dovid B
• [asterisk-users] wrong billsec, when using dial-flag M (was:Answering machine detection after Dial())   Johannes Zweng
• [asterisk-users] Using Modems with Asterisk   Doug Lytle
• [asterisk-users] Using Modems with Asterisk   Jeremy Mann
• [asterisk-users] GotoIf Dialplan inquiry   Lee Jenkins
• [asterisk-users] WAV file best sound quality   Akpome Akpoguma
• [asterisk-users] second dial, force hangup for exit.   Germán Aracil Boned
• [asterisk-users] SPA400 and asterisk   Nick Seraphin
• [asterisk-users] Using Modems with Asterisk   Doug Lytle
• [asterisk-users] Using Modems with Asterisk   Paradise Dove
• [asterisk-users] Digium mailing list server maintenance - Thursday, June 14, 5PM to 8PM CDT   mailman at lists.digium.com
• [asterisk-users] Weird sip registration problem   Anthony Francis
• [asterisk-users] Using Modems with Asterisk   Edwin Lam
• [asterisk-users] Re: Gigabit SIP Phones   Benny Amorsen
• [asterisk-users] Using Modems with Asterisk   Tzafrir Cohen
• [asterisk-users] Addons   David
• [asterisk-users] WAV file best sound quality   Matt
• [asterisk-users] No audio after Dial with G option   Luis Morales
• [asterisk-users] Addons   Atis
• [asterisk-users] CallerID issues   Eric Lubow
• [asterisk-users] Re: [asterisk-dev] CDR changes in Trunk -- Transfers, CDRs, Life, and Everything   Steve Murphy
• [asterisk-users] What is the state of Asterisk Secure Remote Communications?   Alvin Austin
• [asterisk-users] problem starting asterisk, unable to load chan_zap   Doug Lytle
• [asterisk-users] Disconnect tone detection.   Alex Balashov
• [asterisk-users] Re: asterisk-users Digest, Vol 35, Issue 52   Yeruult Dorjsuren
• [asterisk-users] Asterisk Faxing   C F
• [asterisk-users] Re: Slightly OT:CSU on Digium cards, and it's requirement   C F
• [asterisk-users] Asterisk Faxing   Paradise Dove
• [asterisk-users] Re: Slightly OT:CSU on Digium cards, and it's requirement   Erik Anderson
• [asterisk-users] Re: Slightly OT:CSU on Digium cards, and it's requirement   Jon Pounder
• [asterisk-users] Re: Slightly OT:CSU on Digium cards, and it's requirement   Jon Pounder
• [asterisk-users] problem starting asterisk, unable to load chan_zap   Tzafrir Cohen
• [asterisk-users] problem starting asterisk, unable to load chan_zap   Sebastian Reitenbach
• [asterisk-users] Changing the From field in Asterisk email/voicemail   Mark Davies
• [asterisk-users] problem starting asterisk, unable to load chan_zap   Sebastian Reitenbach
• [asterisk-users] problem starting asterisk, unable to load chan_zap   Tzafrir Cohen
• [asterisk-users] Re: asterisk testing - thanx!   Atis
• [asterisk-users] Queue problem   Elmar Haneke
• [asterisk-users] TDM400p and te110p configuration.   Matt Scott
• [asterisk-users] What is the state of Asterisk Secure Remote Communications?   Tim Panton
• [asterisk-users] Qualify renders all SIP peers unreachable   randulo
• [asterisk-users] TDM400p and te110p configuration.   Tzafrir Cohen
• [asterisk-users] Addons   Alexandre VERNIOL
• [asterisk-users] WAV file best sound quality   Akpome Akpoguma
• [asterisk-users] TDM400p and te110p configuration.   Matt Scott
• [asterisk-users] (no subject)   Akpome Akpoguma
• [asterisk-users] Sugar Auto-Dial with Asterisk?   Matt
• [asterisk-users] TDM400p and te110p configuration.   William Moore
• [asterisk-users] Sugar Auto-Dial with Asterisk?   Nuria Fernandez
• [asterisk-users] Adtran feature codes, extensions   Charles Ulrich
• [asterisk-users] Sugar Auto-Dial with Asterisk?   Matt
• [asterisk-users] Sugar Auto-Dial with Asterisk?   EdPimentl
• [asterisk-users] Linksys SPA941   Shad Mortazavi
• [asterisk-users] Re: Slightly OT:CSU on Digium cards, and it's requirement   C F
• [asterisk-users] Re: Slightly OT:CSU on Digium cards, and it's requirement   C F
• [asterisk-users] Re: Slightly OT:CSU on Digium cards, and it's requirement   C F
• [asterisk-users] Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event   Lee Jenkins
• [asterisk-users] ESI Phone System Integration   Jeremy Mann
• [asterisk-users] Sugar Auto-Dial with Asterisk?   Matt
• [asterisk-users] Re: Slightly OT:CSU on Digium cards, and it's requirement   Jon Pounder
• [asterisk-users] Re: Slightly OT:CSU on Digium cards, and it's requirement   Jon Pounder
• [asterisk-users] Re: Slightly OT:CSU on Digium cards, and it's requirement   mail-lists
• [asterisk-users] SIP Options Reply Ignored   randulo
• [asterisk-users] real-time HINTS   Tony Plack
• [asterisk-users] Sending text to a phone that's no in-use ...   Gordon Henderson
• [asterisk-users] Sending text to a phone that's no in-use ...   Rob Schall
• [asterisk-users] ODBC voicemail questions   Kyle Sexton
• [asterisk-users] Sending text to a phone that's no in-use ...   Gordon Henderson
• [asterisk-users] Linksys SPA941   Matt
• [asterisk-users] Linksys SPA941   John Millican
• [asterisk-users] Sending text to a phone that's no in-use ...   Mojo with Horan & Company, LLC
• [asterisk-users] WAV file best sound quality   randulo
• [asterisk-users] Sending text to a phone that's no in-use ...   Jason Parker
• [asterisk-users] Sending text to a phone that's no in-use ...   Mojo with Horan & Company, LLC
• [asterisk-users] Sending text to a phone that's no in-use ...   Dave Fullerton
• [asterisk-users] Sending text to a phone that's no in-use ...   Mojo with Horan & Company, LLC
• [asterisk-users] WAV file best sound quality   Matt
• [asterisk-users] Asterisk GUI   bilal ghayyad
• [asterisk-users] Qualify renders all SIP peers unreachable   Jaswinder Singh
• [asterisk-users] Que on A2Billing   Nitesh Divecha
• [asterisk-users] Re: Slightly OT:CSU on Digium cards, and it's requirement   Nick Seraphin
• [asterisk-users] b410p   Luis José Da Silva González
• [asterisk-users] Asterisk GUI   Erik Anderson
• [asterisk-users] Qualify renders all SIP peers unreachable   randulo
• [asterisk-users] Sending text to a phone that's no in-use ...   Rob Schall
• [asterisk-users] Asterisk GUI   Rob Schall
• [asterisk-users] Re: Slightly OT:CSU on Digium cards, and it's requirement   Alex Balashov
• [asterisk-users] Que on A2Billing   Guillermo Salas M.
• [asterisk-users] Sending text to a phone that's no in-use ...   Mojo with Horan & Company, LLC
• [asterisk-users] question on capacity   Jerry Geis
• [asterisk-users] question on capacity   »Steven Ringwald«
• [asterisk-users] question on capacity   Remco Post
• [asterisk-users] Unicall + MFC/R2 line dropped immediately after connect   Peter Gubis
• [asterisk-users] My Kernel   bilal ghayyad
• [asterisk-users] My Kernel   Remco Post
• [asterisk-users] Crashes with Spandsp, app_rxfax.c, and asterisk 1.4.4   Tzafrir Cohen
• [asterisk-users] My Kernel   Tzafrir Cohen
• [asterisk-users] My Kernel   David Gomillion
• [asterisk-users] Unicall + MFC/R2 line dropped immediately afterconnect   Oscar Carriles
• [asterisk-users] Unicall + MFC/R2 line dropped immediately afterconnect   Oscar Carriles
• [asterisk-users] ODBC voicemail questions   Jared Smith
• [asterisk-users] Sending text to a phone that's no in-use ...   Philipp von Klitzing
• [asterisk-users] Que on A2Billing   Nitesh Divecha
• [asterisk-users] testing   John D. Scott
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• [asterisk-users] My Kernel   Jeff Davis
• [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement   C F
• [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement   C F
• [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement   Alex Balashov
• [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement   Nick Seraphin
• [asterisk-users] Reinvite / one-way media.   Alex Balashov
• [asterisk-users] app_rxfax vs (iaxmodem+hylafax)   Paradise Dove
• [asterisk-users] Sending text to a phone that's no in-use ...   Gordon Henderson
• [asterisk-users] Error: Unable to allocate RTCP socket: Too many open files   Yusuf
• [asterisk-users] hangup during voicemail announcement drops all calls   gincantalupo
• [asterisk-users] My Kernel   Dovid B
• [asterisk-users] Unicall + MFC/R2 line dropped immediately afterconnect   Peter Gubis
• [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement   C F
• [asterisk-users] Run as root?   Malcom Kemp
• [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement   Jeff Davis
• [asterisk-users] g729 codec   Kevin Smith
• [asterisk-users] can ENUMLOOKUP query multiple DNS servers without having to replicate the same code for each server?   rjcarvalho at reit.up.pt
• [asterisk-users] app_rxfax vs (iaxmodem+hylafax)   Doug Lytle
• [asterisk-users] Run as root?   gincantalupo
• [asterisk-users] g729 codec   Tzafrir Cohen
• [asterisk-users] mISDN problem   Ex Vitorino
• [asterisk-users] app_rxfax vs (iaxmodem+hylafax)   Steve Underwood
• [asterisk-users] Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes   Russell Bryant
• [asterisk-users] app_rxfax vs (iaxmodem+hylafax)   Olivier
• [asterisk-users] WaitExten not responding on key presses   Vitell Listmaster
• [asterisk-users] app_rxfax vs (iaxmodem+hylafax)   Tzafrir Cohen
• [asterisk-users] Que on A2Billing   Nitesh Divecha
• [asterisk-users] app_rxfax vs (iaxmodem+hylafax)   Rodrigo Gonzalez
• [asterisk-users] Que on A2Billing   Guillermo Salas M.
• [asterisk-users] Recomender Server specs for 250 con-current calls   Vamsi Pottangi
• [asterisk-users] Transfer caller direct to voicemail   Drew Gibson
• [asterisk-users] Que on A2Billing   Nitesh Divecha
• [asterisk-users] combining AGI with dialplans   james at dades.ca
• [asterisk-users] FXS card with 3-way call, transfer and call waiting.   Paulo Garcia
• [asterisk-users] Que on A2Billing   Guillermo Salas M.
• [asterisk-users] Community PBX?   Kyle Sexton
• [asterisk-users] Run as root?   Remco Post
• [asterisk-users] app_rxfax vs (iaxmodem+hylafax)   Paradise Dove
• [asterisk-users] Community PBX?   Gonzalo Servat
• [asterisk-users] Que on A2Billing   Nitesh Divecha
• [asterisk-users] Where an extension really is (DUNDi woes)   Kyle Sexton
• [asterisk-users] scaling with SMP   Mark Price
• [asterisk-users] Que on A2Billing   Guillermo Salas M.
• [asterisk-users] calling   Carol McGeehon
• [asterisk-users] Que on A2Billing   Nitesh Divecha
• [asterisk-users] Que on A2Billing   Guillermo Salas M.
• [asterisk-users] calling   Dean Collins
• [asterisk-users] Where an extension really is (DUNDi woes)   Anthony Francis
• [asterisk-users] combining AGI with dialplans   Lee Jenkins
• [asterisk-users] calling   Carol McGeehon
• [asterisk-users] Accessing Voicemail(Asterisk) -- Remotely either from Cell or Landline   Deepak Naidu
• [asterisk-users] Asterisk 1.2.19 and 1.4.5 released!   The Asterisk Development Team
• [asterisk-users] Que on A2Billing   Nitesh Divecha
• [asterisk-users] Que on A2Billing   Nitesh Divecha
• [asterisk-users] Asterisk 1.4.anything on FreeBSD?   Bruce Komito
• [asterisk-users] Unicall + MFC/R2 line dropped immediately afterconnect   Steve Underwood
• [asterisk-users] combining AGI with dialplans   james at dades.ca
• [asterisk-users] Asterisk to Panasonic TDA200 with Unicall   Carlos Chavez
• [asterisk-users] Where an extension really is (DUNDi woes)   Kyle Sexton
• [asterisk-users] SOLVED -- Accessing Voicemail(Asterisk) -- Remotely either from Cell or Landline   Deepak Naidu
• [asterisk-users] Asterisk GUI   Paul Hales
• [asterisk-users] Asterisk GUI   Senad Jordanovic
• [asterisk-users] Record CDR in a Oracle database   Tim Panton
• [asterisk-users] Chanspy   Carlos Garcia Mujica
• [asterisk-users] Chanspy   Atis
• [asterisk-users] ivr testing script   Ben Wellborn
• [asterisk-users] (no subject)   Asif Raza
• [asterisk-users] MixMonitor Problem   Asif Raza
• [asterisk-users] Asterisk GUI   Brett Crapser
• [asterisk-users] ipv6 on Asterisk   Jason Ma
• [asterisk-users] Asterisk GUI   Senad Jordanovic
• [asterisk-users] can ENUMLOOKUP query multiple DNS servers without having to replicate the same code for each server?   Remco Post
• [asterisk-users] Asterisk GUI   Tzafrir Cohen
• [asterisk-users] Asterisk GUI   Dean Collins
• [asterisk-users] Convert or listen to .sln file   Andrew Pogrebennyk
• [asterisk-users] Where an extension really is (DUNDi woes)   Remco Post
• [asterisk-users] Convert or listen to .sln file   Atis
• [asterisk-users] Convert or listen to .sln file   Andrew Pogrebennyk
• [asterisk-users] VPN on Asterisk   sdcharly at gmail.com
• [asterisk-users] WAV file best sound quality   randulo
• [asterisk-users] asterisk hang (Critical Response)   Rilawich Ango
• [asterisk-users] Asterisk GUI   Senad Jordanovic
• [asterisk-users] VPN on Asterisk   Dominik Zalewski
• [asterisk-users] VPN on Asterisk   Tim Panton
• [asterisk-users] Mitel 5340 IP Phone   Andy J. Neillans
• [asterisk-users] Asterisk GUI   Tzafrir Cohen
• [asterisk-users] Asterisk Faxing   Kyle Vorster
• [asterisk-users] MixMonitor Problem   Doug Lytle
• [asterisk-users] Transfer caller direct to voicemail   Dovid B
• [asterisk-users] Upgrade cisco SIP phone 7940   Adrian Marsh
• [asterisk-users] MixMonitor Problem   Atis
• [asterisk-users] SIP Peering--call terminated prematurely   Don Kelly
• [asterisk-users] Upgrade cisco SIP phone 7940   Guillermo Salas M.
• [asterisk-users] VPN on Asterisk   Michiel van Baak
• [asterisk-users] Upgrade cisco SIP phone 7940   Kelvin Williams
• [asterisk-users] Upgrade cisco SIP phone 7940   Adrian Marsh
• [asterisk-users] SIP Peering--call terminated prematurely   Don Kelly
• [asterisk-users] SIP Peering--call terminated prematurely   Don Kelly
• [asterisk-users] SIP Peering--call terminated prematurely   Jaswinder Singh
• [asterisk-users] CNAM.   Alex Balashov
• [asterisk-users] CNAM.   Nick Seraphin
• [asterisk-users] CNAM.   Alex Balashov
• [asterisk-users] CNAM.   Nick Seraphin
• [asterisk-users] Asterisk GUI   Paul Hales
• [asterisk-users] Regarding call transfer feature   Priyalatha B
• [asterisk-users] chan problem   Josu Lazkano
• [asterisk-users] chan problem   Tzafrir Cohen
• [asterisk-users] chan problem   Josu Lazkano
• [asterisk-users] Problem In Installing Asterisk on Solaris   Gautam.Yadav at relianceada.com
• [asterisk-users] atxfer attended transfer feature   Vieri
• [asterisk-users] VPN on Asterisk   Biju
• [asterisk-users] asterisk and SAP   nik600
• [asterisk-users] VPN on Asterisk   Michelle Dupuis
• [asterisk-users] VPN on Asterisk   Dominik Zalewski
• [asterisk-users] chan problem   Josu Lazkano
• [asterisk-users] VPN on Asterisk   Jon Pounder
• [asterisk-users] Blind xfer issue -- URGENT!   Jay Moore
• [asterisk-users] VPN on Asterisk   David Gomillion
• [asterisk-users] VPN on Asterisk   Rob Schall
• [asterisk-users] VPN on Asterisk   SIP
• [asterisk-users] VPN on Asterisk   David Gomillion
• [asterisk-users] chan problem   Tzafrir Cohen
• [asterisk-users] chan problem   Josu Lazkano
• [asterisk-users] Blind xfer issue -- URGENT!   Atis
• [asterisk-users] Transfer caller direct to voicemail   Drew Gibson
• [asterisk-users] res_jabber over OpenSSL ready for testing   Philippe Sultan
• [asterisk-users] Blind xfer issue -- URGENT!   Rizwan Hisham
• [asterisk-users] atxfer attended transfer feature   Don Pobanz
• [asterisk-users] asterisk and SAP   Stefan Reuter
• [asterisk-users] Phantom Calls   Lee Jenkins
• [asterisk-users] asterisk and SAP   nik600
• [asterisk-users] atxfer attended transfer feature   Vieri
• [asterisk-users] chan problem   linux at shelob.nl
• [asterisk-users] Monitor recording losing sync   Edgar A. Luna Diaz
• [asterisk-users] chan problem   Bob Chiodini
• [asterisk-users] Phantom Calls   Stephen Bosch
• [asterisk-users] asterisk hang (Critical Response)   Doug
• [asterisk-users] simple dial plan question   achim at itsistem.ro
• [asterisk-users] High availability Asterisk   Mark Phillips
• [asterisk-users] SIP Termination with automatic debit   Douglas Garstang
• [asterisk-users] Phantom Calls   Lee Jenkins
• [asterisk-users] High availability Asterisk   Kristian Kielhofner
• [asterisk-users] SIP Termination with automatic debit   Bob Chiodini
• [asterisk-users] MixMonitor Timestamp problem   Asif Raza
• [asterisk-users] VPN on Asterisk   Remco Barendse
• [asterisk-users] chan problem   Tzafrir Cohen
• [asterisk-users] MixMonitor Timestamp problem   Eric Lubow
• [asterisk-users] MixMonitor Timestamp problem   Savoy, Kevin - Williston, ND
• [asterisk-users] Blind xfer issue -- URGENT!   Jay Moore
• [asterisk-users] chan problem   Tzafrir Cohen
• [asterisk-users] Phantom Calls   Rob Schall
• [asterisk-users] Phantom Calls   Matt
• [asterisk-users] simple dial plan question   Carlos Rojas
• [asterisk-users] VPN on Asterisk   Nuno Vieira - nfsi telecom
• [asterisk-users] atxfer attended transfer feature   Eric "ManxPower" Wieling
• [asterisk-users] no sound with chan_mobile   linux at shelob.nl
• [asterisk-users] VPN on Asterisk   Eric "ManxPower" Wieling
• [asterisk-users] 180 Ringing with SDP   Douglas Garstang
• [asterisk-users] sip <> zap calls choppy, where to setup the jbuffer?   Jay Wilton
• [asterisk-users] Phantom Calls   Lee Jenkins
• [asterisk-users] Asterisk-addons 1.2.7 and 1.4.2 released   The Asterisk Development Team
• [asterisk-users] AGI command   Ronaldo Z. Afonso
• [asterisk-users] How to config SIP blind transfer in extension.conf   Lucian Romi
• [asterisk-users] How to config SIP blind transfer in extension.conf   Alex Balashov
• [asterisk-users] 180 Ringing with SDP   Jared Smith
• [asterisk-users] 180 Ringing with SDP   Alex Balashov
• [asterisk-users] Invalid DTMF detection -- Invalid Extension Bug or issue   Deepak Naidu
• [asterisk-users] AGI command   Martin B. Smith
• [asterisk-users] Need to increase call count   Mike Diehl
• [asterisk-users] Need to increase call count   Luki
• [asterisk-users] passing privacy information through asterisk   Florian Meister
• [asterisk-users] peer timeouts and 489s   Adrian Marsh
• [asterisk-users] make config   Josu Lazkano
• [asterisk-users] Blind xfer issue -- URGENT!   Rizwan Hisham
• [asterisk-users] make config   atik
• [asterisk-users] Play dial tone withou answer   Arjan Kroon
• [asterisk-users] make config   atik
• [asterisk-users] problem with mISDN   Josu Lazkano
• [asterisk-users] 180 Ringing with SDP   Raj Jain
• [asterisk-users] ENUMLOOKUP well succeeded but callee server unreached   Ricardo Carvalho
• [asterisk-users] Play dial tone withou answer   David Boyd
• [asterisk-users] Need to increase call count   Dave Bour
• [asterisk-users] Execute Chanspy   Carlos Garcia Mujica
• [asterisk-users] Execute Chanspy   Carlos Garcia Mujica
• [asterisk-users] Phantom Calls   Vadim Berezniker
• [asterisk-users] Execute ChanSpy   Carlos Garcia Mujica
• [asterisk-users] Execute ChanSpy   William Moore
• [asterisk-users] PhpAgi call generation   nik600
• [asterisk-users] PhpAgi call generation   Rob Schall
• [asterisk-users] Que on A2Billing   Nitesh Divecha
• [asterisk-users] PhpAgi call generation   Christopher Robinson
• [asterisk-users] Phantom Calls   Lee Jenkins
• [asterisk-users] PhpAgi call generation   nik600
• [asterisk-users] PhpAgi call generation   Rob Schall
• [asterisk-users] PhpAgi call generation   Nitesh Divecha
• [asterisk-users] Play dial tone withou answer   Lee Jenkins
• [asterisk-users] Que on A2Billing   Guillermo Salas M.
• [asterisk-users] PhpAgi call generation   Rob Schall
• [asterisk-users] PhpAgi call generation   nik600
• [asterisk-users] PhpAgi call generation   Christopher Robinson
• [asterisk-users] problem with mISDN   Josu Lazkano
• [asterisk-users] Play dial tone withou answer   David Boyd
• [asterisk-users] 180 Ringing with SDP   Douglas Garstang
• [asterisk-users] SIP Termination with automatic debit   Ira
• [asterisk-users] PhpAgi call generation   Christopher Robinson
• [asterisk-users] Asterisk GUI   Tom Rymes
• [asterisk-users] 180 Ringing with SDP   Alex Balashov
• [asterisk-users] Execute ChanSpy   Mojo with Horan & Company, LLC
• [asterisk-users] Asterisk GUI   Senad Jordanovic
• [asterisk-users] Invalid DTMF detection -- Invalid Extension Bug or issue   Doug
• [asterisk-users] Que on A2Billing   Nitesh Divecha
• [asterisk-users] PhpAgi call generation   Nitesh Divecha
• [asterisk-users] Inline record   Adrian Marsh
• [asterisk-users] Debug meetme   Adrian Marsh
• [asterisk-users] 180 Ringing with SDP   Raj Jain
• [asterisk-users] CNAM.   James FitzGibbon
• [asterisk-users] Que on A2Billing   Eric "ManxPower" Wieling
• [asterisk-users] PhpAgi call generation   Lee Jenkins
• [asterisk-users] Inline record   Drew Gibson
• [asterisk-users] Que on A2Billing   John Novack
• [asterisk-users] CNAM.   Al Bochter
• [asterisk-users] Que on A2Billing   Al Bochter
• [asterisk-users] Que on A2Billing   John Novack
• [asterisk-users] Urgent. When the peer returned a 301 forwarded, asterisk thinks it's a local extension.   Lucian Romi
• [asterisk-users] Inline record   Rob Schall
• [asterisk-users] Fwd: Urgent. When the peer returned a 301 forwarded, asterisk thinks it's a local extension.   Lucian Romi
• [asterisk-users] Advice   Duracom Lists
• [asterisk-users] Urgent. When the peer returned a 301 forwarded, asterisk thinks it's a local extension.   Alex Balashov
• [asterisk-users] Advice   Rob Schall
• [asterisk-users] Fwd: Urgent. When the peer returned a 301 forwarded, asterisk thinks it's a local extension.   Lucian Romi
• [asterisk-users] Ex-Girlfriend Logic in 1.4.4   Douglas Garstang
• [asterisk-users] Urgent. When the peer returned a 301 forwarded, asterisk thinks it's a local extension.   Lucian Romi
• [asterisk-users] Fwd: Urgent. When the peer returned a 301 forwarded, asterisk thinks it's a local extension.   BJ Weschke
• [asterisk-users] Fwd: Urgent. When the peer returned a 301 forwarded, asterisk thinks it's a local extension.   Lucian Romi
• [asterisk-users] Que on A2Billing   Nitesh Divecha
• [asterisk-users] How to config SIP blind transfer in extension.conf   Lucian Romi
• [asterisk-users] Que on A2Billing   Al Bochter
• [asterisk-users] Ex-Girlfriend Logic in 1.4.4   Marco Mouta
• [asterisk-users] How to config SIP blind transfer in extension.conf   Lucian Romi
• [asterisk-users] Ex-Girlfriend Logic in 1.4.4   Watkins, Bradley
• [asterisk-users] DTMF detection -- Zaptel   Deepak Naidu
• [asterisk-users] ChanSpy SIP   Ed Nunez
• [asterisk-users] asterisk hang (Critical Response)   Rilawich Ango
• [asterisk-users] WHAT happened to AgentMonitorOutgoing(c) in Asterisk 1.4.5 ??   Roi Stork
• [asterisk-users] Error: Unable to allocate RTCP socket: Too manyopen files   Yusuf
• [asterisk-users] ipv6 on Asterisk   Chris Hills
• [asterisk-users] Inline record   Adrian Marsh
• [asterisk-users] Inline record   Adrian Marsh
• [asterisk-users] mISDN problem   Josu Lazkano
• [asterisk-users] Single ringer phone for incoming calls, that anyone can answer   Tom Lanyon
• [asterisk-users] Inline record   Adrian Marsh
• [asterisk-users] Asterisk GUI   bilal ghayyad
• [asterisk-users] zlib1g   bilal ghayyad
• [asterisk-users] Asterisk GUI   Andrew Latham
• [asterisk-users] Asterisk GUI   Tzafrir Cohen
• [asterisk-users] zlib1g   Tzafrir Cohen
• [asterisk-users] asterisk + mediant 2000   satish patel
• [asterisk-users] Query regarding connecting PABX with Application server   sravi kumar
• [asterisk-users] Agent auto congesting   rachid
• [asterisk-users] Res: Record CDR in a Oracle database   Everton Goularth
• [asterisk-users] ChanSpy SIP   Ed Nuñez
• [asterisk-users] Asterisk RealTime   Nitesh Divecha
• [asterisk-users] Phantom Calls   Dave Miller
• [asterisk-users] hanging up   Julian Lyndon-Smith
• [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy   Jason Ma
• [asterisk-users] Phantom Calls   Lee Jenkins
• [asterisk-users] hanging up   Gabriel Lopes
• [asterisk-users] Phantom Calls   Lee Jenkins
• [asterisk-users] asterisk with mediant 2000 trunk   satish patel
• [asterisk-users] X-Lite problems on basic asterisk setup   Andrew Stewart
• [asterisk-users] ATT: Brian Fertig   Dean Collins
• [asterisk-users] How to Create Custom Context   Nitesh Divecha
• [asterisk-users] How to Create Custom Context   Alex Balashov
• [asterisk-users] X-Lite problems on basic asterisk setup   Rob Schall
• [asterisk-users] Forcing Dial application to skip if called server is unreachable   rjcarvalho at reit.up.pt
• [asterisk-users] Forcing Dial application to skip if called server is unreachable   Alex Balashov
• [asterisk-users] different codec for different extensions   Nasir Iqbal
• [asterisk-users] mISDN problem   Ex Vitorino
• [asterisk-users] Forcing Dial application to skip if called server is unreachable   Ex Vitorino
• [asterisk-users] Asterisk RealTime   Jared Smith
• [asterisk-users] Asterisk GUI   Tom Rymes
• [asterisk-users] Ex-Girlfriend Logic in 1.4.4   Nasir Iqbal
• [asterisk-users] Firewall on AsteriskNow   WipeOut
• [asterisk-users] Dynamically adding Context in dialplan?   Nasir Iqbal
• [asterisk-users] Asterisk GUI   Rob Schall
• [asterisk-users] mISDN problem   Josu Lazkano
• [asterisk-users] Asterisk GUI   Troy Ayers
• [asterisk-users] asterisk hang (Critical Response)   Angel Luis Martinez
• [asterisk-users] X-Lite problems on basic asterisk setup   Andrew Stewart
• [asterisk-users] Gigabit SIP Phones   Dave Bour
• [asterisk-users] zlib1g   bilal ghayyad
• [asterisk-users] Ex-Girlfriend Logic in 1.4.4   Mojo with Horan & Company, LLC
• [asterisk-users] Improving Asterisk's DNS support   Steven
• [asterisk-users] install Asterisk-addons 1.4.2   clive.chan(Alpha Trilogies Networks)
• [asterisk-users] Asterisk 1.2.0 addon Radius   Igor Bonny
• [asterisk-users] zlib1g   Tzafrir Cohen
• [asterisk-users] zlib1g   Al Bochter
• [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql   Khaled Chehab
• [asterisk-users] SIP/IAX2 Phones behind USR 9108 Router   sdcharly at gmail.com
• [asterisk-users] Asterisk config files and #include   Deepak Bhat
• [asterisk-users] Problem with Remote-Hold/MusicOnHold   Gunnar Schaller
• [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql   Khaled Chehab
• [asterisk-users] gtalk - no audio   Koen Van Impe
• [asterisk-users] chan problem   linux at shelob.nl
• [asterisk-users] ipv6 on Asterisk   Tim Panton
• [asterisk-users] Asterisk GUI   Senad Jordanovic
• [asterisk-users] chan problem   Josu Lazkano
• [asterisk-users] chan problem   Tzafrir Cohen
• [asterisk-users] Asterisk config files and #include   Tzafrir Cohen
• [asterisk-users] chan problem   Tzafrir Cohen
• [asterisk-users] mISDN problems   Enrico Pasqualotto
• [asterisk-users] mediant 2000 with asterik configuration   satish patel
• [asterisk-users] chan problem   Josu Lazkano
• [asterisk-users] Asterisk GUI   Senad Jordanovic
• [asterisk-users] gtalk - no audio   demuel at thephinix.org
• [asterisk-users] chan problem   Tzafrir Cohen
• [asterisk-users] mediant 2000 with asterik configuration   Tzafrir Cohen
• [asterisk-users] chan problem   Josu Lazkano
• [asterisk-users] TDM400 one way calls   Matt Scott
• [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql   Khaled Chehab
• [asterisk-users] TDM400 one way calls   Matt Scott
• [asterisk-users] Asterisk config files and #include   Deepak Bhat
• [asterisk-users] chan problem   Josu Lazkano
• [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql   Khaled Chehab
• [asterisk-users] gtalk - no audio   Koen Van Impe
• [asterisk-users] gtalk - no audio   Philippe Sultan
• [asterisk-users] Asterisk config files and #include   Bryan Laird
• [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql   Bryan Laird
• [asterisk-users] chan problem   Tzafrir Cohen
• [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql   Leonardo Kamache (Gmail)
• [asterisk-users] Asterisk config files and #include   Tzafrir Cohen
• [asterisk-users] chan problem   Josu Lazkano
• [asterisk-users] Problem with Remote-Hold/MusicOnHold   Gunnar Schaller
• [asterisk-users] ChanSkype   Kyle Vorster
• [asterisk-users] Asterisk Faxing   Kyle Vorster
• [asterisk-users] Using Queue - Zap problems (PRI)   equis software
• [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql   James Texter
• [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql   Watkins, Bradley
• [asterisk-users] AudioCodes Gateway and Asterisk   Dovid B
• [asterisk-users] ChanSkype   David Gomillion
• [asterisk-users] retreiving callid of call from the dial application   Danish Samad
• [asterisk-users] CDR   Khaled Chehab
• [asterisk-users] Zaptel wct2xxp driver causes LEDs to go black and RED alarm   Jason K. Carter
• [asterisk-users] ENC: Action Originate (Asterisk Manager) X Monitor()   Moacir O. de Souza Junior - Personalsoft Sistemas Ltda.
• [asterisk-users] CDR   Khaled Chehab
• [asterisk-users] identifying what a user pressed to reach my phone   Ryan Stille
• [asterisk-users] AudioCodes Gateway and Asterisk   Atis
• [asterisk-users] identifying what a user pressed to reach my phone   Dave Bour
• [asterisk-users] gtalk - no audio   Joseph Bajin
• [asterisk-users] identifying what a user pressed to reach my phone   Tim Litwiller
• [asterisk-users] TDM800P - zaptel service startup problem   Vidura Senadeera
• [asterisk-users] Improving Asterisk's DNS support   Kristian Kielhofner
• [asterisk-users] Zaptel wct2xxp driver causes LEDs to go black and RED alarm   James Texter
• [asterisk-users] AudioCodes Gateway and Asterisk   Shanon Swafford
• [asterisk-users] Forward to my phones the domain of the CALLERID in incoming URI calls   Ricardo Carvalho
• [asterisk-users] Bug in Ex-Girlfriend logic?   Douglas Garstang
• [asterisk-users] gtalk - no audio   Koen Van Impe
• [asterisk-users] Zaptel wct2xxp driver causes LEDs to go black and RED alarm   Jason K. Carter
• [asterisk-users] Improving Asterisk's DNS support   Darrick Hartman (lists)
• [asterisk-users] Asterisk GUI   Tom Rymes
• [asterisk-users] TDM800P - zaptel service startup problem   Arturo Ochoa
• [asterisk-users] CDR   Bryan Laird
• [asterisk-users] gtalk - no audio   demuel at thephinix.org
• [asterisk-users] gtalk - no audio   demuel at thephinix.org
• [asterisk-users] gtalk - no audio   Philippe Sultan
• [asterisk-users] ATT: Brian Fertig   Andrew Joakimsen
• [asterisk-users] TDM400 one way calls   randulo
• [asterisk-users] gtalk - no audio   Joseph Bajin
• [asterisk-users] identifying what a user pressed to reach my phone   Benny Amorsen
• [asterisk-users] Use of ChanSpy   Carlos Garcia Mujica
• [asterisk-users] Console channels with two sound cards?   Tony Mountifield
• [asterisk-users] Use of ChanSpy   Alex Balashov
• [asterisk-users] Polycoms lose registration and won't re-register   stoffell
• [asterisk-users] gtalk - no audio   demuel at thephinix.org
• [asterisk-users] gtalk - no audio   demuel at thephinix.org
• [asterisk-users] different codec for different extensions   Mojo with Horan & Company, LLC
• [asterisk-users] Blind xfer issue -- URGENT!   Mojo with Horan & Company, LLC
• [asterisk-users] Looking to buy VoIP or Hosting Company   US National Telecom
• [asterisk-users] Blind xfer issue -- URGENT!   Wes Baehr
• [asterisk-users] hotline with Polycom   Klaverstyn, David C
• [asterisk-users] hotline with Polycom   Jonathan Barratt
• [asterisk-users] STDERR in AGI   Ronaldo Z. Afonso
• [asterisk-users] VPN on Asterisk   Nuno Vieira - nfsi telecom
• [asterisk-users] Zaptel wct2xxp driver causes LEDs to go black and RED alarm   Jason K. Carter
• [asterisk-users] Asterisk Faxing   C F
• [asterisk-users] different codec for different extensions   Nasir Iqbal
• [asterisk-users] Asterisk config files and #include   Deepak Bhat
• [asterisk-users] Once Touch Recording   Klaverstyn, David C
• [asterisk-users] solution mediant 2000 with asterik configuration   satish patel
• [asterisk-users] Play dial tone withou answer   Arjan Kroon
• [asterisk-users] qozap and zt_alarm_notify_no_master_change   Vieri
• [asterisk-users] RTCP NTP clock skew detected   Pavel Jezek
• [asterisk-users] chan_features.so / asterisk 1.4.5   Jack
• [asterisk-users] international numbers...   Kevin Withnall
• [asterisk-users] Query   sanchal.singh at alliance-infotech.com
• [asterisk-users] asterisk 0 dial outgoing call   satish patel
• [asterisk-users] Bridged PRI calls - processor involvement?   Steve Hanselman
• [asterisk-users] qozap and zt_alarm_notify_no_master_change   Tzafrir Cohen
• [asterisk-users] asterisk 0 dial outgoing call   ram
• [asterisk-users] Config for TEI parameter   Salvatore
• [asterisk-users] international numbers...   Gary Mensenares
• [asterisk-users] problem with one way audio   Don Briggs
• [asterisk-users] Query   sanchal.singh at alliance-infotech.com
• [asterisk-users] POTS - Incoming Voice or Fax - How to tell?   Gary
• [asterisk-users] Config for TEI parameter   Tzafrir Cohen
• [asterisk-users] Asterisk GUI   Dave Bour
• [asterisk-users] Friday June 22 at 12:30PM EDT Asterisk Users Conference   randulo
• [asterisk-users] Query   Tzafrir Cohen
• [asterisk-users] Asterisk GUI   Steve Kennedy
• [asterisk-users] qozap and zt_alarm_notify_no_master_change   Vieri
• [asterisk-users] got-name   Bill Michaelson
• [asterisk-users] asterisk 0 dial outgoing call   satish patel
• [asterisk-users] Play dial tone withou answer   David Boyd
• [asterisk-users] Asterisk GUI   Tzafrir Cohen
• [asterisk-users] Blind xfer issue -- URGENT!   Jay Moore
• [asterisk-users] Once Touch Recording   Drew Gibson
• [asterisk-users] Query   ram
• [asterisk-users] Query   Steve Totaro
• [asterisk-users] asterisk 0 dial outgoing call   David Boyd
• [asterisk-users] chan_features.so / asterisk 1.4.5   José Luis Ledesma
• [asterisk-users] gtalk - no audio   Philippe Sultan
• [asterisk-users] chan_zap problems   equis software
• [asterisk-users] got-name   Daryl Jones
• [asterisk-users] PhpAgi call generation   Nitesh Divecha
• [asterisk-users] different codec for different extensions   Yusuf
• [asterisk-users] POTS - Incoming Voice or Fax - How to tell?   Alex Balashov
• [asterisk-users] How to config SIP blind transfer in extension.conf   Idris AVCI
• [asterisk-users] problem with one way audio   Lee Jenkins
• [asterisk-users] got-name   Jeff Davis
• [asterisk-users] international numbers...   Dave Bour
• [asterisk-users] searching for compatible servers   Hart Green
• [asterisk-users] Hints   Ken Williams
• [asterisk-users] Asterisk GUI   John Novack
• [asterisk-users] got-name   Luki
• [asterisk-users] gtalk - no audio   demuel at thephinix.org
• [asterisk-users] access to asterisk server since internet   skalli yassir
• [asterisk-users] Binding to multiple ports in sip.conf   R. Raja
• [asterisk-users] Does Early Media have to be Ulaw?   Douglas Garstang
• [asterisk-users] international numbers...   Benny Amorsen
• [asterisk-users] Blind xfer issue -- URGENT!   Wes Baehr
• [asterisk-users] Audio going one way for a few seconds during the call   Zeeshan Zakaria
• [asterisk-users] got-name   Jeff Davis
• [asterisk-users] Ring/Off-hook in strange state 6   Alex Mcdowell
• [asterisk-users] searching for compatible servers   Gary G. Hendershot
• [asterisk-users] got-name   Nick Seraphin
• [asterisk-users] inband DTMF for g729   Gary Chen
• [asterisk-users] Nuance Buys Tegic from AOL for $265m   Dean Collins
• [asterisk-users] H.323 IP Phones and H.323 Traffic   bilal ghayyad
• [asterisk-users] Binding to multiple addresses   Jordan Novak
• [asterisk-users] inband DTMF for g729   Matthew Fredrickson
• [asterisk-users] Friday June 22 at 12:30PM EDT Asterisk Users Conference   randulo
• [asterisk-users] 1.4.5   Ed Nuñez
• [asterisk-users] got-name   Jonathan Creasy
• [asterisk-users] inband DTMF for g729   Gary Chen
• [asterisk-users] inband DTMF for g729   Ed Nuñez
• [asterisk-users] Ex-Girlfriend Logic in 1.4.4   Anthony Francis
• [asterisk-users] asterisk-users Digest, Vol 35, Issue 81   Bill Michaelson
• [asterisk-users] inband DTMF for g729   Kristian Kielhofner
• [asterisk-users] Polycom 301 - Problem with AMI Originated Calls   Lee Jenkins
• [asterisk-users] Ring/Off-hook in strange state 6   Daniel Hazelbaker
• [asterisk-users] gtalk - no audio   Philippe Sultan
• [asterisk-users] Asterisk GUI   Anthony Francis
• [asterisk-users] STDERR in AGI   Anthony Francis
• [asterisk-users] PhpAgi call generation   Anthony Francis
• [asterisk-users] FAX over T1   Joe acquisto
• [asterisk-users] Query   Deepak Naidu
• [asterisk-users] inband DTMF for g729   Jon Weisman
• [asterisk-users] FAX over T1   Carlos Chavez
• [asterisk-users] FAX over T1   C F
• [asterisk-users] Does Early Media have to be Ulaw?   Kristian Kielhofner
• [asterisk-users] install Asterisk-addons 1.4.2   Ed Nunez
• [asterisk-users] modules loading   clive.chan(Alpha Trilogies Networks)
• [asterisk-users] Asternic Flash panel   John Faubion
• [asterisk-users] DTMF Problem with one sip trunk and two carriers   jk at bingoconsulting.com
• [asterisk-users] Single ringer phone for incoming calls, that anyone can answer   Gordon Henderson
• [asterisk-users] Asterisk config files and #include   Deepak Bhat
• [asterisk-users] asterisk 0 dial outgoing call   satish patel
• [asterisk-users] IVR question for asterisk   satish patel
• [asterisk-users] Dynamic DUNDi weight   Andre Wangler
• [asterisk-users] chan_features.so / asterisk 1.4.5   Carsten Bock
• [asterisk-users] FAX over T1   Doug Lytle
• [asterisk-users] IAX client USB phone   Ronaldo Z. Afonso
• [asterisk-users] IVR question for asterisk   Dean Collins
• [asterisk-users] TAS test equipment manuals   Steve Underwood
• [asterisk-users] IAX client USB phone   Dean Collins
• [asterisk-users] IVR question for asterisk   Tzafrir Cohen
• [asterisk-users] Zaptel Compilation   bilal ghayyad
• [asterisk-users] IAX client USB phone   Gordon Henderson
• [asterisk-users] Zaptel Compilation Error   bilal ghayyad
• [asterisk-users] chan_features.so / asterisk 1.4.5   John Faubion
• [asterisk-users] Zaptel Compilation Error   Tzafrir Cohen
• [asterisk-users] Zaptel Compilation   F6HQZ
• [asterisk-users] TAS test equipment manuals   F6HQZ
• [asterisk-users] IAX client USB phone   Michael Graves
• [asterisk-users] Dynamic DUNDi weight   Jared Smith
• [asterisk-users] STDERR in AGI   Jared Smith
• [asterisk-users] FAX over T1   Marco Mouta
• [asterisk-users] Call Path Optimization   Ronaldo Z. Afonso
• [asterisk-users] OT: Digg post   Dean Collins
• [asterisk-users] Single ringer phone for incoming calls, that anyone can answer   Tom Lanyon
• [asterisk-users] Single ringer phone for incoming calls, that anyone can answer   mitcheloc
• [asterisk-users] TAS test equipment manuals   F6HQZ
• [asterisk-users] Zaptel Compilation   F6HQZ
• [asterisk-users] problem with one way audio   Jason Backshall
• [asterisk-users] Binding to multiple addresses   Jaswinder Singh
• [asterisk-users] problem with one way audio   Zeeshan Zakaria
• [asterisk-users] inband DTMF for g729   Gang Chen
• [asterisk-users] inband DTMF for g729   Darrick Hartman (lists)
• [asterisk-users] inband DTMF for g729   Julio Arruda
• [asterisk-users] inband DTMF for g729   Andres
• [asterisk-users] Nokia N95 + Dial Plan   Nitesh Divecha
• [asterisk-users] inband DTMF for g729   Joshua Colp
• [asterisk-users] inband DTMF for g729   Darrick Hartman (lists)
• [asterisk-users] Use of ChanSpy   Oscar Carriles
• [asterisk-users] problem with one way audio   Zeeshan Zakaria
• [asterisk-users] Nokia N95 + Dial Plan   Benny Amorsen
• [asterisk-users] Use of ChanSpy   Jaswinder Singh
• [asterisk-users] H.323 IP Phones and H.323 Traffic   Dovid B
• [asterisk-users] AudioCodes Gateway and Asterisk   Dovid B
• [asterisk-users] AudioCodes Gateway and Asterisk   Dovid B
• [asterisk-users] How to Create Custom Context   Antonio José dos Santos Brandão
• [asterisk-users] Single ringer phone for incoming calls, that anyone can answer   William McCloskey
• [asterisk-users] Zaptel Compilation   bilal ghayyad
• [asterisk-users] Zaptel Compilation   Steve Totaro
• [asterisk-users] Faktortel in Sydney outage   Dean Collins
• [asterisk-users] Provisioning Linksys WIP330 phones   Michelle Dupuis
• [asterisk-users] Best wifi IP phone for asterisk   Michelle Dupuis
• [asterisk-users] Query   sanchal.singh at alliance-infotech.com
• [asterisk-users] Audio going one way for a few seconds during thecall   Jason Backshall
• [asterisk-users] IAX client USB phone   Anselm Martin Hoffmeister
• [asterisk-users] chan problem   linux at shelob.nl
• [asterisk-users] asterisk not able to hear calling party ring sound   satish patel
• [asterisk-users] Xorcom Bri 4 Port USB   Nathan Dennis
• [asterisk-users] Query   Tzafrir Cohen
• [asterisk-users] Xorcom Bri 4 Port USB   Tzafrir Cohen
• [asterisk-users] g729 problem   ram
• [asterisk-users] Forcing Dial application to skip if called server is unreachable   Ricardo Carvalho
• [asterisk-users] Best wifi IP phone for asterisk   Marco Mouta
• [asterisk-users] ChanSkype   Hugo Miguel de Almeida Teixeira Picao
• [asterisk-users] outging select   Josu Lazkano
• [asterisk-users] Hi ability solution   voip crazy
• [asterisk-users] Best wifi IP phone for asterisk   Benny Amorsen
• [asterisk-users] g729 problem   Karl J. Vesterling
• [asterisk-users] Hi ability solution   Steve Totaro
• [asterisk-users] g729 problem   ram
• [asterisk-users] outging select   Steve Totaro
• [asterisk-users] inband DTMF for g729   Gary Chen
• [asterisk-users] Rining 180 and 183   satish patel
• [asterisk-users] Ring/Off-hook in strange state 6   Alex Mcdowell
• [asterisk-users] outging select   Josu Lazkano
• [asterisk-users] Hi ability solution   voip crazy
• [asterisk-users] Hi ability solution   Steve Totaro
• [asterisk-users] Hi ability solution   Senad Jordanovic
• [asterisk-users] Best wifi IP phone for asterisk   Marcus Franke
• [asterisk-users] Nokia N95 + Dial Plan   Nitesh Divecha
• [asterisk-users] asterisk-users Digest, Vol 35, Issue 89   jr at kahari.com
• [asterisk-users] 1.4.5   Vadim Berezniker
• [asterisk-users] Rining 180 and 183   Jared Smith
• [asterisk-users] Call Path Optimization   Matthew Fredrickson
• [asterisk-users] Best wifi IP phone for asterisk   Michelle Dupuis
• [asterisk-users] asterisk-users Digest, Vol 35, Issue 89   Dean Collins
• [asterisk-users] two channels, each dropping into the same context, different behavior.   Ryan Goldberg
• [asterisk-users] callback and bridge problem   Adam KOSA
• [asterisk-users] Set a global queue policy   Alvaro Parres
• [asterisk-users] Hi ability solution   Steve Totaro
• [asterisk-users] Rining 180 and 183   satish patel
• [asterisk-users] Threading troubles 1.4.5 & IAX2-> SIP (FreeBSD specific??)   Hendrik Visage
• [asterisk-users] international numbers...   Eric "ManxPower" Wieling
• [asterisk-users] Best wifi IP phone for asterisk   Benny Amorsen
• [asterisk-users] Hi ability solution   Senad Jordanovic
• [asterisk-users] IAX client USB phone   Michiel van Baak
• [asterisk-users] two channels, each dropping into the same context, different behavior.   Ryan Goldberg
• [asterisk-users] Res: Record CDR in a Oracle database   Tim Panton
• [asterisk-users] Asterisk 1.4.5, Cisco 7960, call dropped when sip client put on hold/transfer   falz
• [asterisk-users] Outbound proxy setting with outbound proxy port in sip.conf   Lucian Romi
• [asterisk-users] Asterisk 1.4.5, Cisco 7960, call dropped when sip client put on hold/transfer   Carlos Chavez
• [asterisk-users] Does outboundproxyport still work in 1.4.4   Lucian Romi
• [asterisk-users] Asterisk 1.4.5, Cisco 7960, call dropped when sip client put on hold/transfer   falz
• [asterisk-users] BRI Layer 2 status   asterisk
• [asterisk-users] Rining 180 and 183   Mojo with Horan & Company, LLC
• [asterisk-users] Transfer Call to Cell Phone   OCOSA ListAcc
• [asterisk-users] Help. Help. Help. How to make outbound proxy and host URI with different port?   Lucian Romi
• [asterisk-users] two channels, each dropping into the same context, different behavior.   Andres Paglayan
• [asterisk-users] BRI Layer 2 status   Tzafrir Cohen
• [asterisk-users] four ringing and hangup with error   satish patel
• [asterisk-users] Dynamic DUNDi weight support in * - HELP!   Andre Wangler
• [asterisk-users] Problems with ChanIsAvail always return status 0   Alvaro Parres
• [asterisk-users] asterisk-users Digest, Vol 35, Issue 89   John Faubion
• [asterisk-users] Ring the second line when 1st line is busy   Deepak Naidu
• [asterisk-users] Best wifi IP phone for asterisk   Nick Seraphin
• [asterisk-users] AstPligg   lenz
• [asterisk-users] Transfer Call to Cell Phone   John Faubion
• [asterisk-users] AstPligg   Mark Phillips
• [asterisk-users] Best wifi IP phone for asterisk   Marco Mouta
• [asterisk-users] asterisk-users Digest, Vol 35, Issue 91   jr at kahari.com
• [asterisk-users] CDR Records "s" as dst   Troy - Purple Oranges
• [asterisk-users] Problems with ChanIsAvail always return status 0   Jared Smith
• [asterisk-users] Threading troubles 1.4.5 & IAX2-> SIP (FreeBSD specific??)   Jared Smith
• [asterisk-users] Dynamic DUNDi weight support in * - HELP!   Jared Smith
• [asterisk-users] Spy a specific Channel   Carlos Garcia Mujica
• [asterisk-users] Transfer Call to Cell Phone   OCOSA ListAcct
• [asterisk-users] Modification of Caller ID based on context   arkda
• [asterisk-users] Fax Throughput   Don Kelly
• [asterisk-users] Best wifi IP phone for asterisk   Andrew Joakimsen
• [asterisk-users] CDR changes in 1.4.5 are confusing   Mail list
• [asterisk-users] call transfer problem   satish patel
• [asterisk-users] Inexpensive Layer 3 Switch?   Marty Mastera
• [asterisk-users] Nagios asterisk monitoring   Kenny Kant
• [asterisk-users] Inexpensive Layer 3 Switch?   James FitzGibbon
• [asterisk-users] CDR Records "s" as dst   Jaswinder Singh
• [asterisk-users] AstPligg   lenz
• [asterisk-users] Transfer Call to Cell Phone   John Faubion
• [asterisk-users] Threading troubles 1.4.5 & IAX2-> SIP (FreeBSD specific??)   Hendrik Visage
• [asterisk-users] Best wifi IP phone for asterisk   Hendrik Visage
• [asterisk-users] Rining 180 and 183   Andrew Joakimsen
• [asterisk-users] Inexpensive Layer 3 Switch?   John Faubion
• [asterisk-users] zaptel 1.2.18 and HPEC   Paul Hales
• [asterisk-users] Asterisk + Legacy PBX   Marc Patino Gómez
• [asterisk-users] CDR Records "s" as dst   Troy - Purple Oranges
• [asterisk-users] realtime_extensions   Pezhman Lali
• [asterisk-users] Best wifi IP phone for asterisk   Nick Seraphin
• [asterisk-users] asterisk-users Digest, Vol 35, Issue 92   jr at kahari.com
• [asterisk-users] VPN technology for snom 370   Hirosh Dabui
• [asterisk-users] Bridging two PSTN calls   Doug Zingel
• [asterisk-users] Query   sanchal.singh at alliance-infotech.com
• [asterisk-users] Test Message   Gary
• [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql   didier
• [asterisk-users] Digium TE120P setup problem [was: Re: Query]   Tzafrir Cohen
• [asterisk-users] call fail from audiocode to sip trunk   satish patel
• [asterisk-users] Fax Throughput   Lee Howard
• [asterisk-users] chan_features.so / asterisk 1.4.5   Jack
• [asterisk-users] AstPligg   Alex Robar
• [asterisk-users] Best wifi IP phone for asterisk (LINKSYS SUPPORT QUALITY)   Michelle Dupuis
• [asterisk-users] No CID on Zaps - TDM400   Eric Estes
• [asterisk-users] chan_features.so / asterisk 1.4.5   Joshua Colp
• [asterisk-users] AstPligg   Jon Weisman
• [asterisk-users] Transfer Call to Cell Phone   OCOSA ListAcct
• [asterisk-users] AstPligg   Alex Robar
• [asterisk-users] Multi port IAX Gateway   Mike Hammett
• [asterisk-users] Inexpensive Layer 3 Switch?   Jerry Jones
• [asterisk-users] rcf2833 DTMF broken in asterisk SIP channel?   tracinet
• [asterisk-users] Fax Throughput   Matthew Fredrickson
• [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy   tracinet
• [asterisk-users] chan_features.so / asterisk 1.4.5   Jack
• [asterisk-users] Transfer Call to Cell Phone   John Faubion
• [asterisk-users] Inexpensive Layer 3 Switch?   Marty Mastera
• [asterisk-users] More FAX over T1   Joe acquisto
• [asterisk-users] AstPligg   lenz
• [asterisk-users] Bridging two PSTN calls   Don Pobanz
• [asterisk-users] Inexpensive Layer 3 Switch?   David Gomillion
• [asterisk-users] CDR Records "s" as dst   Eric "ManxPower" Wieling
• [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy   Eric "ManxPower" Wieling
• [asterisk-users] SpectraLink SVP protocol support in asterisk   Michelle Dupuis
• [asterisk-users] Problems with ChanIsAvail always return status 0   Alvaro Parres
• [asterisk-users] Inexpensive Layer 3 Switch?   John Faubion
• [asterisk-users] More FAX over T1   David Gomillion
• [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy   Ed Nuñez
• [asterisk-users] Inexpensive Layer 3 Switch?   Steve Totaro
• [asterisk-users] Ring/Off-hook in strange state 6   Alex Mcdowell
• [asterisk-users] More FAX over T1   Joe acquisto
• [asterisk-users] Slip Events   Jon Weisman
• [asterisk-users] Ring/Off-hook in strange state 6   Bruce Reeves
• [asterisk-users] More FAX over T1   Lacy Moore - Aspendora
• [asterisk-users] Bad Echo between SIP calls   Mojo with Horan & Company, LLC
• [asterisk-users] Ring/Off-hook in strange state 6   Eric "ManxPower" Wieling
• [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy   Jason Ma
• [asterisk-users] Inexpensive Layer 3 Switch?   Marty Mastera
• [asterisk-users] More FAX over T1   David Gomillion
• [asterisk-users] More FAX over T1   Joe acquisto
• [asterisk-users] Inexpensive Layer 3 Switch?   David Gomillion
• [asterisk-users] kore dump   Ed Nuñez
• [asterisk-users] TE412 / HPDL380G5 / * 1.4 / CentOS 4.5 Experience   James FitzGibbon
• [asterisk-users] kore dump   John Faubion
• [asterisk-users] kore dump   Vadim Berezniker
• [asterisk-users] Transfer Call to Cell Phone   OCOSA ListAcc
• [asterisk-users] kore dump   Jared Smith
• [asterisk-users] More FAX over T1   Lacy Moore - Aspendora
• [asterisk-users] More FAX over T1   Joe acquisto
• [asterisk-users] kore dump   Eric Lubow
• [asterisk-users] Cisco 7941 localized menus with SIP firmware   Olivier
• [asterisk-users] Ring/Off-hook in strange state 6   Alex Mcdowell
• [asterisk-users] Cisco 7941 localized menus with SIP firmware   Steve Kennedy
• [asterisk-users] Fax Throughput   Don Kelly
• [asterisk-users] Cisco 7941 localized menus with SIP firmware   Olivier
• [asterisk-users] More FAX over T1   David Gomillion
• [asterisk-users] kore dump   J. Oquendo
• [asterisk-users] zaptel 1.2.18 and HPEC   Kevin P. Fleming
• [asterisk-users] No such host error from SIP for non-peer configuration.   Lucian Romi
• [asterisk-users] Bridging two PSTN calls   Jared Smith
• [asterisk-users] No such host error from SIP for non-peer configuration.   Lucian Romi
• [asterisk-users] Provisioning Linksys WIP330 phones   Shanon Swafford
• [asterisk-users] kore dump   Luki
• [asterisk-users] Modification of Caller ID based on context   Matthew Brothers
• [asterisk-users] AudioCodes Gateway and Asterisk   Shanon Swafford
• [asterisk-users] Asterisk to Cisco 2600 GW DTMF Not Working   JR Richardson
• [asterisk-users] Modification of Caller ID based on context   arkda
• [asterisk-users] kore dump   Mojo with Horan & Company, LLC
• [asterisk-users] Ring the second line when 1st line is busy   Deepak Naidu
• [asterisk-users] access to asterisk server since internet   Mojo with Horan & Company, LLC
• [asterisk-users] Asterisk to Cisco 2600 GW DTMF Not Working   Mark Phillips
• [asterisk-users] Need to increase call count   Mike Diehl
• [asterisk-users] access to asterisk server since internet   Don Kelly
• [asterisk-users] Nokia N95 + Dial Plan   Kevin Withnall
• [asterisk-users] Module '***.so' did not register itself during load   Igor Bonny
• [asterisk-users] Transfer Call to Cell Phone   Ryan Goldberg
• [asterisk-users] Asterisk GUI   Paul Hales
• [asterisk-users] kore dump   Luki
• [asterisk-users] Ring the second line when 1st line is busy   Rob Townley
• [asterisk-users] Transfer Call to Cell Phone   John Faubion
• [asterisk-users] Urgent. When the peer returned a 301 forwarded, asterisk thinks it's a local extension.   John Faubion
• [asterisk-users] Modification of Caller ID based on context   Eric "ManxPower" Wieling
• [asterisk-users] AudioCodes Gateway and Asterisk   Dovid B
• [asterisk-users] Fwd: problem with one way audio   Vidura Senadeera
• [asterisk-users] Zap dialling issues   Nathan Dennis
• [asterisk-users] Missing 'init keys' command   Jonathan Unai Marquez
• [asterisk-users] IAX trunking using a different port   Ronaldo
• [asterisk-users] minibrowser for each snom phone   Hirosh Dabui
• [asterisk-users] ISDN data-call question   Steve Davies
• [asterisk-users] Self Calling test   Dave Bour
• [asterisk-users] Zap dialling issues   Tzafrir Cohen
• [asterisk-users] Missing 'init keys' command   Jared Smith
• [asterisk-users] Wait to numbers   Josu Lazkano
• [asterisk-users] Help with IAX Trunk   Arun Kumar
• [asterisk-users] Round Robin SIP peers?   Mark Phillips
• [asterisk-users] Problems compiling Asterisk 1.4.5   equis software
• [asterisk-users] Multi port IAX Gateway   William Moore
• [asterisk-users] Problems compiling Asterisk 1.4.5   Drew Gibson
• [asterisk-users] Problems compiling Asterisk 1.4.5   Jared Smith
• [asterisk-users] Help with IAX Trunk   Jared Smith
• [asterisk-users] Wait to numbers   Jared Smith
• [asterisk-users] kore dump   Ed Nuñez
• [asterisk-users] Problems compiling Asterisk 1.4.5   equis software
• [asterisk-users] kore dump   William Moore
• [asterisk-users] Fax Throughput   Matthew Fredrickson
• [asterisk-users] kore dump   Jared Smith
• [asterisk-users] problem with one way audio   Lee Jenkins
• [asterisk-users] Wait to numbers   Josu Lazkano
• [asterisk-users] Help with IAX Trunk   Arun Kumar
• [asterisk-users] Problems compiling Asterisk 1.4.5   Anthony Francis
• [asterisk-users] Asterisk+squid   rozsa
• [asterisk-users] Asterisk+squid   Anthony Francis
• [asterisk-users] Asterisk to Cisco 2600 GW DTMF Not Working, Working now   JR Richardson
• [asterisk-users] Asterisk+squid   David Gomillion
• [asterisk-users] kore dump   John Faubion
• [asterisk-users] Problems compiling Asterisk 1.4.5   equis software
• [asterisk-users] Nuance Buys Tegic from AOL for $265m   Steve Underwood
• [asterisk-users] inband DTMF for g729   Steve Underwood
• [asterisk-users] Using MSAccess to dial on a Zap line   Jason Martin
• [asterisk-users] Self Calling test   Mojo with Horan & Company, LLC
• [asterisk-users] Customized Ring Tone   GNUbie
• [asterisk-users] Round Robin SIP peers?   Alex Balashov
• [asterisk-users] inband DTMF for g729   Alex Balashov
• [asterisk-users] Has anyone sucessful Asterisk to an Avaya IP phone   Bob Gibson
• [asterisk-users] Has anyone sucessful Asterisk to an Avaya IP phone   Bob Gibson
• [asterisk-users] Customized Ring Tone   Alexander Lopez
• [asterisk-users] Using MSAccess to dial on a Zap line   Alex Balashov
• [asterisk-users] OpenSer/Asterisk PBX solution   Bob Gibson
• [asterisk-users] Asterisk+squid   Kristian Kielhofner
• [asterisk-users] QueueMetrics 1.4 released today   Lenz
• [asterisk-users] Asterisk+squid   Eric "ManxPower" Wieling
• [asterisk-users] .call file   Nitesh Divecha
• [asterisk-users] Error While Calling AGI   Nitesh Divecha
• [asterisk-users] .call file   Jerry Geis
• [asterisk-users] Has anyone sucessful Asterisk to an Avaya IP phone   Bryan M. Johns
• [asterisk-users] Cisco 7941 localized menus with SIP firmware   Greg Oliver
• [asterisk-users] .call file   Nitesh Divecha
• [asterisk-users] .call file   Jerry Geis
• [asterisk-users] North American voice BRI - Informal survey   Stephen Bosch
• [asterisk-users] North American voice BRI - Informal survey   Greg Oliver
• [asterisk-users] North American voice BRI - Informal survey   Stephen Bosch
• [asterisk-users] North American voice BRI - Informal survey   Jay R. Ashworth
• [asterisk-users] North American voice BRI - Informal survey   Joe Greco
• [asterisk-users] Bypass local dialplan and redirect INVITE   Lucian Romi
• [asterisk-users] North American voice BRI - Informal survey   James Van Vleet
• [asterisk-users] Voicestick / i2telecom.com   Huw Richards
• [asterisk-users] Using MSAccess to dial on a Zap line   C F
• [asterisk-users] Voicestick / i2telecom.com   Dave Bour
• [asterisk-users] North American voice BRI - Informal survey   Stephen Bosch
• [asterisk-users] Voicestick / i2telecom.com   Huw Richards
• [asterisk-users] Transfer Call to Cell Phone   OCOSA ListAcc
• [asterisk-users] Transfer Call to Cell Phone   OCOSA ListAcc
• [asterisk-users] North American voice BRI - Informal survey   Joe Greco
• [asterisk-users] Xorcom Bri 4 Port USB   Nathan Dennis
• [asterisk-users] Using MSAccess to dial on a Zap line   BJ Weschke
• [asterisk-users] Customized Ring Tone   GNUbie
• [asterisk-users] Any difference using * with Centos i386 and x86_64 ?   Andre Gustavo Lomonaco
• [asterisk-users] Ring/Off-hook in strange state 6   Alex Mcdowell
• [asterisk-users] voicemail.conf serveremail   Patrick Pfeifer
• [asterisk-users] callback and bridge problem   Adam KOSA
• [asterisk-users] Call transfer feature   satish patel
• [asterisk-users] Updated Manual for Asterisk 1.4.x   GNUbie
• [asterisk-users] Calls audio stops with latest Gigaset C450IP firmware + voicemail   gincantalupo
• [asterisk-users] Ring/Off-hook in strange state 6   Dimitri Volski
• [asterisk-users] registering Asterisk on SIP/Nortel MCS thing   Kate Kretz
• [asterisk-users] ooh323 hang up after the call is answered   Kate Kretz
• [asterisk-users] Work   Paul Hales
• [asterisk-users] fail to load modules   clive.chan(Alpha Trilogies Networks)
• [asterisk-users] error while compiling asterisk-1.2.19   Goke Aruna
• [asterisk-users] voicemail.conf serveremail   Dave Bour
• [asterisk-users] CDR and call transfer   Grigoriy Puzankin
• [asterisk-users] Missing 'init keys' command   Jonathan Unai Marquez
• [asterisk-users] Fax passthrough howto codec upspeed   Ivoc
• [asterisk-users] Voicestick / i2telecom.com   Huw Richards
• [asterisk-users] Asterisk 1.4.5 Inserting Random Digits in Dialed Number!   Russell Brown
• [asterisk-users] registering Asterisk on SIP/Nortel MCS server   Kate Kretz
• [asterisk-users] error while compiling asterisk-1.2.19   Tzafrir Cohen
• [asterisk-users] error while compiling asterisk-1.2.19   Goke Aruna
• [asterisk-users] Cisco 7941 localized menus with SIP firmware   Olivier
• [asterisk-users] error while compiling asterisk-1.2.19   Tzafrir Cohen
• [asterisk-users] Error While Calling AGI   Nitesh Divecha
• [asterisk-users] FXS channel bank   pixiesfr
• [asterisk-users] FXS channel bank   Jerry Jones
• [asterisk-users] Using MSAccess to dial on a Zap line   Martin Smith
• [asterisk-users] CDR and call transfer   Grigoriy Puzankin
• [asterisk-users] Caller ID Spoofing to be banned in the USA   Dean Collins
• [asterisk-users] Query   sanchal.singh at alliance-infotech.com
• [asterisk-users] E1 not coming up   Alexander Zielke
• [asterisk-users] Cisco 7941 localized menus with SIP firmware   Greg Oliver
• [asterisk-users] Query   David Gomillion
• [asterisk-users] Asterisk 1.4.5 Inserting Random Digits in Dialed Number!   Eric "ManxPower" Wieling
• [asterisk-users] Updated Manual for Asterisk 1.4.x   Eric "ManxPower" Wieling
• [asterisk-users] Query   Victor Toofic
• [asterisk-users] RTCP NTP Clock skew   John Millican
• [asterisk-users] E1 not coming up   Matthew Fredrickson
• [asterisk-users] FXS channel bank   Alex Balashov
• [asterisk-users] Updated Manual for Asterisk 1.4.x   GNUbie
• [asterisk-users] Using MSAccess to dial on a Zap line   C F
• [asterisk-users] Linking Asterisk with another SIP PBX (or SIP Softswitch)   bilal ghayyad
• [asterisk-users] Caller ID Spoofing to be banned in the USA   J. Oquendo
• [asterisk-users] Call transfer feature   Lee Jenkins
• [asterisk-users] Updated Manual for Asterisk 1.4.x   Drew Gibson
• [asterisk-users] Asterisk + hinting presence + macro   Rob Schall
• [asterisk-users] RTCP NTP Clock skew   Jared Smith
• [asterisk-users] Asterisk + hinting presence + macro   Jaswinder Singh
• [asterisk-users] Avoided deadlock for '0x864e70', 10 retries!   ram
• [asterisk-users] Asterisk + hinting presence + macro   Rob Schall
• [asterisk-users] Xorcom Bri 4 Port USB   Tzafrir Cohen
• [asterisk-users] voicemail.conf serveremail   Stephen Bosch
• [asterisk-users] Avaya IP Office DTMF Issue   Jon Farmer
• [asterisk-users] Asterisk and IPv6   Bent Bagger
• [asterisk-users] network routing   Ed Nuñez
• [asterisk-users] network routing   ~Russell
• [asterisk-users] Updated Manual for Asterisk 1.4.x   Eric "ManxPower" Wieling
• [asterisk-users] Asterisk + hinting presence + macro   Eric "ManxPower" Wieling
• [asterisk-users] Shared Extension Appearance   Mike Ryan
• [asterisk-users] Robo Dialer   NedTel NedTel
• [asterisk-users] Anyone who can do live video feed to co-host asterisk "show" next week?   randulo
• [asterisk-users] CDR Log analizer software   Mark Coccimiglio
• [asterisk-users] Asterisk + hinting presence + macro   Rob Schall
• [asterisk-users] network routing   Ed Nuñez
• [asterisk-users] setup multiple phones for 1 extension   Ryan Stille
• [asterisk-users] network routing   ~Russell
• [asterisk-users] setup multiple phones for 1 extension   Gustavo Hernandez
• [asterisk-users] setup multiple phones for 1 extension   tracinet
• [asterisk-users] North American voice BRI - Informal survey   Benny Amorsen
• [asterisk-users] network routing   Watkins, Bradley
• [asterisk-users] CDR Log analizer software   Ivan Cetta
• [asterisk-users] RTCP NTP Clock skew   John Millican
• [asterisk-users] North American voice BRI - Informal survey   Jon Pounder
• [asterisk-users] setup multiple phones for 1 extension   randulo
• [asterisk-users] North American voice BRI - Informal survey   Jeremy Mann
• [asterisk-users] Cisco 7941 localized menus with SIP firmware   Dan Austin
• [asterisk-users] network routing   Ed Nuñez
• [asterisk-users] North American voice BRI - Informal survey   Stephen Bosch
• [asterisk-users] setup multiple phones for 1 extension   Ryan Stille
• [asterisk-users] North American voice BRI - Informal survey   Jon Pounder
• [asterisk-users] North American voice BRI - Informal survey   Tom
• [asterisk-users] network routing   Joseph Bajin
• [asterisk-users] Agent Channel SIP transfer   Marlon Dutra
• [asterisk-users] Customized Ring Tone   Dimitri Volski
• [asterisk-users] Modification of Caller ID based on context   arkda
• [asterisk-users] SPA-2100 Distinctive Ring   Matt Putnam
• [asterisk-users] Customized Ring Tone   GNUbie
• [asterisk-users] SPA-2100 Distinctive Ring   Luki
• [asterisk-users] Linking Asterisk with another SIP PBX (or SIP Softswitch)   Noah Miller
• [asterisk-users] Shared Extension Appearance   Russell Bryant
• [asterisk-users] Asterisk and IPv6   Russell Bryant
• [asterisk-users] Query   Deepak Naidu
• [asterisk-users] Asterisk + hinting presence + macro   Eric "ManxPower" Wieling
• [asterisk-users] FAX over T1   Andres Paglayan
• [asterisk-users] Asterisk + hinting presence + macro   Jaswinder Singh
• [asterisk-users] E1 not coming up   Alexander Zielke
• [asterisk-users] Cisco 7941 localized menus with SIP firmware   Pavel Jezek
• [asterisk-users] Kirk IP600 V3 DECT Wireless server   Garth van Sittert
• [asterisk-users] How many number of parallel calls can make through asterisk   Santosh S Kumar
• [asterisk-users] How many number of parallel calls can make through asterisk   Yusuf
• [asterisk-users] asterisk call unique id in dialplan   nik600
• [asterisk-users] nway call   Keshav K.
• [asterisk-users] Fwd: Call Wainting dysfunctions   Lorenzo Grosselli
• [asterisk-users] Linking Asterisk with another SIP PBX (or SIP Softswitch)   bilal ghayyad
• [asterisk-users] Query   sanchal.singh at alliance-infotech.com
• [asterisk-users] MOH question w/Cisco 79xx phones   Gary
• [asterisk-users] asterisk call unique id in dialplan   Alexander Lopez
• [asterisk-users] FAX over T1   Steve Totaro
• [asterisk-users] .call file   Kevin Smith
• [asterisk-users] MOH question w/Cisco 79xx phones   Bill Gibbs
• [asterisk-users] DUNDi problem: offline peers still in request EID/EID_DIRECT field?   Andre Wangler
• [asterisk-users] Asterisk 1.4 Warnnings   Umar Draz
• [asterisk-users] features.conf / DTMF / automon hell   Danny Brown
• [asterisk-users] Asterisk 1.4 Warnnings   Martin Smith
• [asterisk-users] Zaptel Compilation   bilal ghayyad
• [asterisk-users] Linking Asterisk with another SIP PBX (or SIP Softswitch)   Mojo with Horan & Company, LLC
• [asterisk-users] Asterisk + hinting presence + macro   Anthony Francis
• [asterisk-users] v1.4.x ready yet?   shadowym
• [asterisk-users] Voice Mail not Receive   Asif Raza
• [asterisk-users] .call file   Paul
• [asterisk-users] Asterisk + hinting presence + macro   Rob Schall
• [asterisk-users] Music on hold 1.2   Ed Nuñez
• [asterisk-users] v1.4.x ready yet?   Bruce Reeves
• [asterisk-users] Music on hold 1.2   Steve Davies
• [asterisk-users] kore dump   Ed Nuñez
• [asterisk-users] Music on hold - 1.4.5   Ade Vickers
• [asterisk-users] Problem getting a Perl script to run   Robert A. Rawlinson
• [asterisk-users] FAX over T1   Andres Paglayan
• [asterisk-users] TE420 PCI Express Card   Andres
• [asterisk-users] Linking Asterisk with another SIP PBX   bilal ghayyad
• [asterisk-users] asterisk call unique id in dialplan   nik600
• [asterisk-users] Problem getting a Perl script to run   Mojo with Horan & Company, LLC
• [asterisk-users] Asterisk + hinting presence + macro   Anthony Francis
• [asterisk-users] Problem getting a Perl script to run   Robert A. Rawlinson
• [asterisk-users] Asterisk 1.2.20 and 1.4.6 released   The Asterisk Development Team
• [asterisk-users] awful list delays: 4 days!   Lenz
• [asterisk-users] v1.4.x ready yet?   Jaswinder Singh
• [asterisk-users] TE420 PCI Express Card   Russell Bryant
• [asterisk-users] awful list delays: 4 days!   Andres Paglayan
• [asterisk-users] TE420 PCI Express Card   Andres
• [asterisk-users] awful list delays: 4 days!   Anthony Francis
• [asterisk-users] awful list delays: 4 days!   Mojo with Horan & Company, LLC
• [asterisk-users] v1.4.x ready yet?   Eric "ManxPower" Wieling
• [asterisk-users] FW: fail to load modules   clive.chan(Alpha Trilogies Networks)
• [asterisk-users] FW: fail to load modules   Russell Bryant
• [asterisk-users] kore dump   Russell Bryant
• [asterisk-users] Fwd: Problems with zap following 1.4.6 install   hugolivude
• [asterisk-users] Fwd: Problems with zap following 1.4.6 install   Russell Bryant
• [asterisk-users] Cisco 7970G line buttons   lists at infoway.net
• [asterisk-users] Kirk IP600 V3 DECT Wireless server   F6HQZ
• [asterisk-users] Query   Keshav K.
• [asterisk-users] Cisco 7970G line buttons   Bill Hackensack
• [asterisk-users] awful list delays: 4 days!   lenz
• [asterisk-users] AEL + Realitme?   Christian Svensson
• [asterisk-users] Asterisk 1.4.6 Fedora 7 configure error   Bill Maidment
• [asterisk-users] Zaptel Compilation   Doug Lytle
• [asterisk-users] Rining 180 and 183   Pavel Jezek
• [asterisk-users] Fwd: Problems with zap following 1.4.6 install   hugolivude
• [asterisk-users] Fwd: Problems with zap following 1.4.6 install   hugolivude
• [asterisk-users] Asterisk 1.4.5 : Use I386 ou X86_64   Andre Gustavo Lomonaco
• [asterisk-users] Zaptel Compilation   Keshav K.
• [asterisk-users] 1.4.5   Keshav K.
• [asterisk-users] Missing 'init keys' command   Russell Bryant
• [asterisk-users] mISDN Fritz Passive Card   Manuel
• [asterisk-users] 1.4.5   Russell Bryant
• [asterisk-users] .call file   Kevin Smith
• [asterisk-users] Zaptel Compilation   Tzafrir Cohen
• [asterisk-users] Polycom echo problem   Jordan Novak
• [asterisk-users] 1.4.5   fneira
• [asterisk-users] Polycom echo problem   Zeeshan Zakaria
• [asterisk-users] G729 , upgrade asterisk   AL Daei
• [asterisk-users] G729 , upgrade asterisk   Darryl Dunkin
• [asterisk-users] G729 , upgrade asterisk   Russell Bryant
• [asterisk-users] Polycom echo problem   Lee Jenkins