June 2005 Archives by date

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Starting: Wed Jun 1 00:21:34 MST 2005
Ending: Thu Jun 30 23:37:41 MST 2005
Messages: 4569

• [Asterisk-Users] newbie with kphone and asterisk   Sukardi Shahdan
• [Asterisk-Users] Phone always busy after caller hangup   stevanus
• [Asterisk-Users] Phone always busy after caller hangup   Terry H. Gilsenan
• [Asterisk-Users] Built-In Transfer Questions   Gavin Hamill
• [Asterisk-Users] Dynamic IAX Server   chawki hammoud
• [Asterisk-Users] Phone always busy after caller hangup   Kulbir Saini
• [Asterisk-Users] When to use 'Answer' and when NOT to...   Chris Coulthurst
• [Asterisk-Users] IVR Load   WipeOut
• [Asterisk-Users] Hardware questions   Aitor
• **POSSIBLE SPAM** [Asterisk-Users] AreskiCC - DOES IT REALLY WORK??????   David Choo
• [Asterisk-Users] UPS rating for SOHO asterisk box   Jean-Michel Hiver
• [Asterisk-Users] Unreliable DTMF detection with DISA on incoming Zap channel on bristuffed * and GSM gateway   Robert Rozman
• [Asterisk-Users] BT101 new firmware problem (1.0.6.3)   Elwin Andriol
• [Asterisk-Users] UPS rating for SOHO asterisk box   Terry H. Gilsenan
• [Asterisk-Users] Problems hanging up PSTN line   db_nz at paradise.net.nz
• [Asterisk-Users] Problems hanging up PSTN line   Terry H. Gilsenan
• [Asterisk-Users] UPS rating for SOHO asterisk box   Jean-Michel Hiver
• [Asterisk-Users] Problems hanging up PSTN line   Mike Price
• [Asterisk-Users] send and receive MMS   Yannick Daronnat
• [Asterisk-Users] debugging zap channel   robert.brown01 at blueyonder.co.uk
• [Asterisk-Users] Problems hanging up PSTN line   db_nz at paradise.net.nz
• R: R: R: R: [Asterisk-Users] AT-320 + supervised transfer   Giordano Grandis
• [Asterisk-Users] hang up a SIP channel   Lee Lee
• R: R: R: R: [Asterisk-Users] AT-320 + supervised transfer   Gavin Hamill
• [Asterisk-Users] Problem with codec negotiation   Mark Dutton
• [Asterisk-Users] Ztdummy usage   Tzafrir Cohen
• [Asterisk-Users] Launching an application from within Asterisk   Paulo
• [Asterisk-Users] pbx -> fiber -> network media converter -> wifi -> network media converter -> fiber -> pbx ???   Rich Adamson
• [Asterisk-Users] Ztdummy usage   Andreas Sikkema
• R: R: R: R: R: [Asterisk-Users] AT-320 + supervised transfer   Giordano Grandis
• [Asterisk-Users] Sipura 2000 behind NAT issue, Vonage is working   Michael George
• [Asterisk-Users] AT-320 + supervised transfer   Gavin Hamill
• R: [Asterisk-Users] AT-320 + supervised transfer   Giordano Grandis
• [Asterisk-Users] UPS rating for SOHO asterisk box   Rich Adamson
• R: [Asterisk-Users] AT-320 + supervised transfer   Gavin Hamill
• [Asterisk-Users] Sipura 2000 behind NAT issue, Vonage is working   David Hajek
• [Asterisk-Users] IVR Load   Mohamed A. Gombolaty
• [Asterisk-Users] asterisk x PROLIANT ML 150 G2 SATA   James Sizemore
• [Asterisk-Users] UPS rating for SOHO asterisk box   Daryl G. Jurbala
• [Asterisk-Users] gnugk   Micko
• [Asterisk-Users] gnugk   Peter Valkov
• R: R: [Asterisk-Users] AT-320 + supervised transfer   Giordano Grandis
• [Asterisk-Users] Dynamic IAX Server   Wiley Siler
• [Asterisk-Users] MOH Jittery Voice   Wiley Siler
• R: R: [Asterisk-Users] AT-320 + supervised transfer   Gavin Hamill
• [Asterisk-Users] FW: TellMe pay-as-you-go? - UPDATE   Dean Collins
• [Asterisk-Users] Sipura 2000 behind NAT issue, Vonage is working   Rich Adamson
• [Asterisk-Users] Asterisk Google API applications - $4500 bounties available   Dean Collins
• [Asterisk-Users] SIP Soft Video phone for Asterisk usage   Nardis Dome
• [Asterisk-Users] pbx -> fiber -> network media converter -> wifi -> network media converter -> fiber -> pbx ???   Ben Dugdale
• [Asterisk-Users] pbx -> fiber -> network media converter -> wifi -> network media converter -> fiber -> pbx ???   Ben Dugdale
• [Asterisk-Users] Setting up a TDM   Hugo Barra
• [Asterisk-Users] Large installation with Asterisk   richard Coco
• [Asterisk-Users] MOH Jittery Voice   usman at user.iphonica.net
• [Asterisk-Users] Pass-through   Adam Vocks
• [Asterisk-Users] [q] About chan_misdn, latest mISDNuser and asterisk cvs   Rus V. Brushkoff
• [Asterisk-Users] Pass-through   Adam Vocks
• [Asterisk-Users] Pass-through   Alexander Lopez
• [Asterisk-Users] Pass-through   Dustin Wildes
• [Asterisk-Users] Pass-through   Dave Weis
• R: R: R: [Asterisk-Users] AT-320 + supervised transfer   Giordano Grandis
• [Asterisk-Users] Pass-through   Adam Vocks
• [Asterisk-Users] Segmentation Fautl / Core Dump with G.729   Jorge Alayon
• [Asterisk-Users] ARESKICC - Another issue   robson at cariocanet.com
• [Asterisk-Users] Last of the servers forsale cheap   Preston Garrison
• [Asterisk-Users] rxfax problems - cont.   Marcin Kuczera
• [Asterisk-Users] VoiPSupply Dot Com   Robert Goodyear
• [Asterisk-Users] astapi memory errors?   Dean Collins
• [Asterisk-Users] DTMF not working   Aitor
• [Asterisk-Users] HELP Cisco - can't find P0S3-07-4-00.sbn   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] Astcc does not work - no repeat metering   C W Nel
• [Asterisk-Users] Pass-through   Alexander Lopez
• [Asterisk-Users] HELP Cisco - can't find P0S3-07-4-00.sbn   Mailing List
• [Asterisk-Users] Suppress "Missed Calls" 7960 SIP   Robert Goodyear
• [Asterisk-Users] pbx -> fiber -> network media converter -> wifi -> network media converter -> fiber -> pbx ???   Bill Ford
• [Asterisk-Users] Fax and codecs preferences to PSTN   René Mayorga
• [Asterisk-Users] list down?   Dean Collins
• [Asterisk-Users] Unreliable DTMF detection with DISA on incomingZap channel on bristuffed * and GSM gateway   Marcelo Sosa Lugones
• [Asterisk-Users] TDM400P Channels stop answering after some time .   Sandeep A.S
• [Asterisk-Users] VoiPSupply Dot Com   Race Vanderdecken
• [Asterisk-Users] Dell SC1425 and TE110P   Oswaldo Arratia
• [Asterisk-Users] ARESKICC - Another issue   Julius Igugu
• [Asterisk-Users] HELP Cisco - can't find P0S3-07-4-00.sbn   Robert Goodyear
• [Asterisk-Users] list down?   Gregory Junker
• [Asterisk-Users] asterisk compatible, hot swappable PRI card   Race Vanderdecken
• [Asterisk-Users] Pass-through   BJ Weschke
• [Asterisk-Users] tellme hiring VXML   Dean Collins
• [Asterisk-Users] Alternate DID   Asterisk
• [Asterisk-Users] 99% cpu on asterisk with chan_unicall and low traffic   Andres Maduro
• [Asterisk-Users] Pri restarting randomly (TE110P or TE405P)   Niklas Larsson
• [Asterisk-Users] HELP Cisco - can't find P0S3-07-4-00.sbn   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] list of settings   Andres Paglayan
• [Asterisk-Users] TDM400P vs SIP3000 x2   Andres Paglayan
• [Asterisk-Users] Cannot receive incoming calls via ISDN   Igor Colombi
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Scott Wolfe
• [Asterisk-Users] A Way to Write DTMF Digits as text to CDR?   PA
• [Asterisk-Users] voice-coloring with asterisk   Script Head
• [Asterisk-Users] RFC2833 & firewall problems? (16-byte UDP packets)   Bryan Field-Elliot
• [Asterisk-Users] ARESKICC - Another issue   robson at cariocanet.com
• [Asterisk-Users] Asterisk@Home 1.1b1 has been released   Kanuri, Seshu (Company IT)
• [Asterisk-Users] A Way to Write DTMF Digits as text to CDR?   pbx at itsngroup.com
• [Asterisk-Users] SetGroup CheckGroup   Joseph
• [Asterisk-Users] Cannot find module (NET-SNMP-EXTEND-MIB)   Michael Stearne
• [Asterisk-Users] Pass-through   Adam Vocks
• [Asterisk-Users] Astcc does not work - no repeat metering   C W Nel
• [Asterisk-Users] SNOM 360 extension lights   Ross Kevlin
• [Asterisk-Users] Asterisk@Home 1.1b1 has been released   Tim Litwiller
• [Asterisk-Users] voice-coloring with asterisk   Michael Loftis
• [Asterisk-Users] Broadvoice - Customer feedback   Sean Kennedy
• [Asterisk-Users] chan_zap.c error   Ed Kenny
• [Asterisk-Users] Asterisk@Home 1.1b1 has been released   Wiley Siler
• [Asterisk-Users] wrong numbers message   Luis Diaz
• [Asterisk-Users] Asterisk@Home 1.1b1 has been released   Wiley Siler
• [Asterisk-Users] Broadvoice - Customer feedback   Dan Perik
• [Asterisk-Users] Broadvoice - Customer feedback   trixter http://www.0xdecafbad.com
• [Asterisk-Users] 1.0.8 Release Candidate   Russell Bryant
• [Asterisk-Users] FW: TellMe pay-as-you-go? - UPDATE   Adam Goryachev
• [Asterisk-Users] Inject Audio into Existing Call   Rod Bacon
• [Asterisk-Users] 1.0.8 Release Candidate   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Broadvoice - Customer feedback   Rusty Shackelford
• [Asterisk-Users] Broadvoice - Customer feedback   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Asterisk@Home 1.1b1 has been released   Oswaldo Arratia
• [Asterisk-Users] Areskicc v2 login issues   Alexandre Charles
• [Asterisk-Users] Newbie Question: HOWTO make outgoing call on SIP account from internal extensions?   Steve
• [Asterisk-Users] Re: Obtaining Cisco Firmware painlessly andLEGITIMATELY?   Ryan Finnesey
• [Asterisk-Users] rxfax problems - cont.   Steve Underwood
• [Asterisk-Users] voice-coloring with asterisk   Steve Underwood
• [Asterisk-Users] Broadvoice - Customer feedback   Ronald Wiplinger
• [Asterisk-Users] Newbie Question: HOWTO make outgoing call on SIP account from internal extensions?   Ronald Wiplinger
• [Asterisk-Users] Built-In Transfer Questions   Paul Hales
• [Asterisk-Users] list of settings   Luis Diaz
• [Asterisk-Users] 1.0.8 Release Candidate   Matt Riddell
• [Asterisk-Users] RFC2833 & firewall problems? (16-byte UDP packets)   Matt Riddell
• [Asterisk-Users] Broadvoice - Customer feedback   Luki
• [Asterisk-Users] IVR Load   Luki
• [Asterisk-Users] CVS HEAD won't compile for me   Jeff Heath
• [Asterisk-Users] Broadvoice - Customer feedback   JD Austin
• [Asterisk-Users] Does Asterisk Realtime require the use of CVS HEAD ???   Jeff Heath
• [Asterisk-Users] Reccomendations for connecting to 3-4 PSTN lines?   Ivan Fetch
• [Asterisk-Users] Realtime+IAX2 and RSA   kleis-asterisk-dev at tiscali.it
• [Asterisk-Users] does asterisk work with other processors   Listacc
• [Asterisk-Users] Broadvoice - Customer feedback   Jerry Geis
• [Asterisk-Users] IAX2 analog telephone adapter   Dinesh
• [Asterisk-Users] MGC on asterisk   shjaszs
• [Asterisk-Users] Supervised/Attended transfers   Jamie
• [Asterisk-Users] Fax and codecs preferences to PSTN   Obaid Siddiqui
• [Asterisk-Users] Re: Obtaining Cisco Firmware painlessly and LEGITIMATELY?   asterisk at marnock.net
• [Asterisk-Users] Supervised/Attended transfers (working, but more Qs)   Jamie Carl
• [Asterisk-Users] Issue with Not Capturing All Key Presses   Michael Stearne
• [Asterisk-Users] Does Asterisk Realtime require the use of CVS HEAD ???   Michael Stearne
• [Asterisk-Users] Legacy PBX -> * -> Voip Calls problems   Scott Wolfe
• [Asterisk-Users] A Way to Write DTMF Digits as text to CDR?   Michael Stearne
• [Asterisk-Users] wrong numbers message   Michael Stearne
• [Asterisk-Users] IAX2 analog telephone adapter   Samy Antoun
• [Asterisk-Users] Reccomendations for connecting to 3-4 PSTN lines?   C F
• [Asterisk-Users] pbx -> fiber -> network media converter -> wifi -> network media converter -> fiber -> pbx ???   Ben Dugdale
• [Asterisk-Users] RFC2833 & firewall problems? (16-byte UDP packets)   Bryan Field-Elliot
• [Asterisk-Users] Incoming and Outgoing   Bill Madeira
• [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server   Samy Antoun
• [Asterisk-Users] SetGroup CheckGroup   C F
• [Asterisk-Users] VoiPSupply Dot Com   C F
• [Asterisk-Users] HELP Cisco - can't find P0S3-07-4-00.sbn   Omar Sabek
• [Asterisk-Users] Karl   C F
• [Asterisk-Users] Newbie Question: HOWTO make outgoing call on SIP account from internal extensions?   Steve
• [Asterisk-Users] 1.0.8 Release Candidate   Michael Stearne
• [Asterisk-Users] RTP Read too short   VoIP Newbie
• [Asterisk-Users] Voice recognition application - VoIP/Open Source   end1r
• [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server   steve szmidt
• [Asterisk-Users] Does Asterisk Realtime require the use of CVS HEAD ???   Matthew Boehm
• [Asterisk-Users] Realtime+IAX2 and RSA   Matthew Boehm
• [Asterisk-Users] Voice recognition application - VoIP/Open Source   Luis Diaz
• [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server   Rod Bacon
• [Asterisk-Users] RFC2833 & firewall problems? (16-byte UDP packets)   Matt Riddell
• [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server   Samy Antoun
• [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server   Samy Antoun
• [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server   Kristian Kielhofner
• [Asterisk-Users] CRM integration (was RE: CallerID)   Adnan Ahmed
• [Asterisk-Users] 4+ Port FXS Analog Device   Waldo Rubinstein
• [Asterisk-Users] SNOM 360 extension lights   Olle E. Johansson
• [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server   Samy Antoun
• [Asterisk-Users] 4+ Port FXS Analog Device   Matthew H. Franz
• [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server   Kristian Kielhofner
• [Asterisk-Users] CVS HEAD won't compile for me   Olle E. Johansson
• [Asterisk-Users] Pricing for DS3000P   Nathan
• [Asterisk-Users] SIP or IAX   Sandeep A.S
• [Asterisk-Users] SIP or IAX   Olle E. Johansson
• [Asterisk-Users] chan_capi + mISDN + Fritz PTP   Ralf Schlatterbeck
• [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server   Tony Mountifield
• [Asterisk-Users] 4+ Port FXS Analog Device   VoIP Newbie
• [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server   Tzafrir Cohen
• [Asterisk-Users] MOH Jittery Voice   usman at user.iphonica.net
• [Asterisk-Users] send and receive MMS   Klaus-Peter Junghanns
• [Asterisk-Users] Newbie :Call Forwarding problem   Mohamed A. Gombolaty
• [Asterisk-Users] Cisco 7960s and skinny   Yusuf Iqbal
• [Asterisk-Users] Newbie :Call Forwarding problem   Peter Bowyer
• [Asterisk-Users] Will my CPU/RAM be sufficient?   Marco Trucchi
• [Asterisk-Users] Newbie :Call Forwarding problem   Mohamed A. Gombolaty
• [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server   Ben Buxton
• [Asterisk-Users] How to connect to Asterisk to IPTEL.ORG   Alex Piqueras
• [Asterisk-Users] Newbie :Call Forwarding problem   Peter Bowyer
• [Asterisk-Users] handytone 486   Betül Gözlükoğlu
• [Asterisk-Users] handytone 486   Olle E. Johansson
• [Asterisk-Users] trunk timing on 2.6.x   Raul Elizondo (wizardteam)
• [Asterisk-Users] SetGroup CheckGroup   Joseph
• [Asterisk-Users] Script to test channel bank   Chris Mason (Lists)
• [Asterisk-Users] H323 trunk with cisco gatekeeper   Mahmoud
• [Asterisk-Users] a simple call to my girlfriend   Hendrik Wouters
• [Asterisk-Users] Astricon Europe :: Tutorial Agenda now published   Olle E. Johansson
• [Asterisk-Users] Newbie :Call Forwarding problem   Mohamed A. Gombolaty
• [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server   C. Hatton Humphrey
• [Asterisk-Users] Re: Obtaining Cisco Firmware painlessly and LEGITIMATELY?   Tony Hoyle
• [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server   Mike Dent
• [Asterisk-Users] Call Meeting VS Call Confrence   Mohamed A. Gombolaty
• [Asterisk-Users] VoiPSupply Dot Com   Matt
• [Asterisk-Users] VoiPSupply Dot Com   Matt
• [Asterisk-Users] a simple call to my girlfriend   Administrator TOOTAI
• [Asterisk-Users] gsm call-hunting [OT]   Eric Smith - Fruitcom
• [Asterisk-Users] Will my CPU/RAM be sufficient?   Roy Sigurd Karlsbakk
• [Asterisk-Users] does asterisk work with other processors   Tom Fanning
• [Asterisk-Users] Reccomendations for connecting to 3-4 PSTN lines?   Rich Adamson
• [Asterisk-Users] Call Meeting VS Call Confrence   Peter Svensson
• [Asterisk-Users] Cisco 7960s and skinny   Julien Goodwin
• [Asterisk-Users] CRM integration (was RE: CallerID)   Patrick
• [Asterisk-Users] TE11OP -> Mitel 200Sx??   Geoff Manning
• [Asterisk-Users] 4+ Port FXS Analog Device   Waldo Rubinstein
• [Asterisk-Users] 4+ Port FXS Analog Device   Waldo Rubinstein
• [Asterisk-Users] bluetooth headset/handsfree   Fredrik Chabot
• [Asterisk-Users] SIP or IAX   steve szmidt
• [Asterisk-Users] Pricing for DS3000P   Andrew Latham
• [Asterisk-Users] 4+ Port FXS Analog Device   Daniel Nyström
• [Asterisk-Users] 4+ Port FXS Analog Device   Andrew Latham
• [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server   steve szmidt
• [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server   Dave Cotton
• [Asterisk-Users] IAX2 analog telephone adapter   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server   pbx at itsngroup.com
• [Asterisk-Users] asterisk like modems access server   Edgardo Lust
• [Asterisk-Users] 4+ Port FXS Analog Device   VoIP Newbie
• [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server   steve szmidt
• [Asterisk-Users] connecting to nortel CS1000 (half way there)   Jerry Geis
• [Asterisk-Users] Does Debian Bristuffed Asterisk work ignore Beronet cards ?   Robert Rozman
• [Asterisk-Users] trunk timing on 2.6.x   Steve Kann
• [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server   Dave Cotton
• [asterisk-users] Reccomendations for connecting to 3-4 PSTN lines?   Ivan Fetch
• [Asterisk-Users] Outgoing Calls via chan_capi   Matthias Böhm
• [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server   Kristian Kielhofner
• [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server   Dave Cotton
• [Asterisk-Users] Asterisk@Home 1.1b1 has been released   Wiley Siler
• [Asterisk-Users] Fax and codecs preferences to PSTN   René Mayorga
• [Asterisk-Users] connect to SIP trunk getting unable to create/find channel   Jerry Geis
• [Asterisk-Users] FW: Help with Kpn e1 settings please   asterisk at truescot.net
• [Asterisk-Users] asterisk like modems access server   Rich Adamson
• [Asterisk-Users] 4+ Port FXS Analog Device   Waldo Rubinstein
• R: [Asterisk-Users] AT-320 + supervised transfer   Kanuri, Seshu (Company IT)
• [Asterisk-Users] trunk timing on 2.6.x   Raul Elizondo (wizardteam)
• [Asterisk-Users] CVS HEAD won't compile for me   chawki hammoud
• [Asterisk-Users] ProSLIC 3210 version 2 is too old.   Hugh L. Johnson
• [Asterisk-Users] Outgoing Calls via chan_capi   Matthias Böhm
• [Asterisk-Users] VoiPSupply Dot Com   The VoIP Connection
• [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server   steve szmidt
• [Asterisk-Users] Replacing SIP Trunking With IAX Trunking   Sandeep A.S
• [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received   Nick Barnes
• [Asterisk-Users] 2 incoming lines and Asterisk@home...   Francois Meehan
• [Asterisk-Users] bison/flex version warning   Mike M
• [Asterisk-Users] Replacing SIP Trunking With IAX Trunking   Wiley Siler
• [Asterisk-Users] 2 incoming lines and Asterisk@home...   Wiley Siler
• [Asterisk-Users] 2 incoming lines and Asterisk@home...   Wiley Siler
• [Asterisk-Users] Will my CPU/RAM be sufficient?   Luki
• [Asterisk-Users] ProSLIC 3210 version 2 is too old.   Hugh L. Johnson
• [Asterisk-Users] asterisk on internet sip phone behind nat - does someone even have this working   Lance Grover
• [Asterisk-Users] asterisk on internet sip phone behind nat - doessomeone even have this working   Wiley Siler
• [Asterisk-Users] VoiPSupply Dot Com   Neal Walton
• [Asterisk-Users] asterisk on internet sip phone behind nat - doessomeone even have this working   Wiley Siler
• [Asterisk-Users] Asterisk Exitstatus of 127?   Nathan Pralle
• [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] VoiPSupply Dot Com   Wiley Siler
• [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] ProSLIC 3210 version 2 is too old.   Rich Adamson
• [Asterisk-Users] 2 incoming lines and Asterisk@home...   Francois Meehan
• [Asterisk-Users] HELP Cisco - can't find P0S3-07-4-00.sbn   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] 2 incoming lines and Asterisk@home...   Wiley Siler
• [Asterisk-Users] Asterisk connecting to nortel CS 1000 as sip trunk Need help with final piece (incoming call) outgoing works.   Jerry Geis
• [Asterisk-Users] Re: Reccomendations for connecting to 3-4 PSTN lines?   Ivan Fetch
• [Asterisk-Users] Does Debian Bristuffed Asterisk work ignore Beronet cards ?   Emanuele Pucciarelli
• [Asterisk-Users] compile asterisk   Kamran Ahmad
• [Asterisk-Users] asterisk on internet sip phone behind nat - doessomeone even have this working   Lance Grover
• [Asterisk-Users] a simple call to my girlfriend   C F
• [Asterisk-Users] Call Manager & Asterisk for VM - MWI not working   info at beprojects.com
• [Asterisk-Users] VoiPSupply Dot Com   The VoIP Connection
• [Asterisk-Users] asterisk on internet sip phone behind nat - doessomeone even have this working   Lance Grover
• [Asterisk-Users] Asterisk hangup for no reason (FXO)   Joel Duffield
• [Asterisk-Users] application sdp message and not answering call   Jerry Geis
• [Asterisk-Users] a simple call to my girlfriend   Tom Fanning
• [Asterisk-Users] bison/flex version warning   Michael Stearne
• [Asterisk-Users] compile asterisk   Juan Pablo Abuyeres
• [Asterisk-Users] bison/flex version warning   Andrew Latham
• [Asterisk-Users] UPS rating for SOHO asterisk box   Walt Reed
• [Asterisk-Users] a simple call to my girlfriend   Yiannis Costopoulos
• [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server   Ed Greenberg
• [Asterisk-Users] asterisk on internet sip phone behind nat - does someone even have this working   Olle E. Johansson
• [Asterisk-Users] asterisk on internet sip phone behind nat - doessomeone even have this working   Olle E. Johansson
• [Asterisk-Users] ProSLIC 3210 version 2 is too old.   Rich Adamson
• [Asterisk-Users] DID Routing over SIP   Asterisk VoIP
• [Asterisk-Users] a simple call to my girlfriend   Robert Goodyear
• [Asterisk-Users] a simple call to my girlfriend   Adam Vocks
• [Asterisk-Users] 2 incoming lines and Asterisk@home...   Asterisk at Home
• [Asterisk-Users] Host Authentication Problems   Dave Kettmann
• [Asterisk-Users] asterisk sipura and g726 codec   Philip Fleischer
• [Asterisk-Users] Does Debian Bristuffed Asterisk work ignoreBeronet cards ?   Robert Rozman
• [Asterisk-Users] asterisk sipura and g726 codec   Trevor Peirce
• [Asterisk-Users] IAX encrytion   Håkan Källberg
• [Asterisk-Users] How to ensure that software echo cancellation is on?   Don Murray
• [Asterisk-Users] Asterisk hangup for no reason (FXO)   Moises Silva
• [Asterisk-Users] a simple call to my girlfriend   Adrian A
• [Asterisk-Users] Does Debian Bristuffed Asterisk work ignoreBeronet cards ?   Peer Oliver Schmidt
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Steve
• [Asterisk-Users] How to ensure that software echo cancellation is on?   jjones at quiddesign.com
• [Asterisk-Users] How to ensure that software echo cancellation is on?   Don Murray
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Kevin Bockman
• [Asterisk-Users] How to disable Digium card ?   Ronald Wiplinger
• [Asterisk-Users] what about dCap certification?   Jordi Clopés
• [Asterisk-Users] Announce: Asterisk virtual configuration   snacktime
• [Asterisk-Users] a simple call to my girlfriend   Michael Graves
• [Asterisk-Users] a simple call to my girlfriend   Mojo with Horan & Company, LLC
• [Asterisk-Users] How to disable Digium card ?   Andrew Latham
• [Asterisk-Users] Announce: Asterisk virtual configuration   Andrew Latham
• [Asterisk-Users] Pricing for DS3000P   izo
• [Asterisk-Users] Announce: Asterisk virtual configuration   Sean Cook
• [Asterisk-Users] bison/flex version warning   Mike M
• [Asterisk-Users] iax went away   Michael George
• [Asterisk-Users] Announce: Asterisk virtual configuration   snacktime
• [Asterisk-Users] Zaptel not found error during modprobe   Pudenz, Duane
• [Asterisk-Users] chan_sip.c:886 retrans_pkt: Maximum retries exceeded   asterisk at marnock.net
• [Asterisk-Users] 1.0.8 Release Candidate   Greg Boehnlein
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 17   Nguyen Trung Tin
• [Asterisk-Users] Announce: Asterisk virtual configuration   Andrew Latham
• [Asterisk-Users] Zaptel not found error during modprobe   Andrew Latham
• [Asterisk-Users] Asterisk RealTime Voicemail Not Working   Michael Stearne
• [Asterisk-Users] voip provider request   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Pricing for DS3000P   Dean Collins
• [Asterisk-Users] IAX2 and Queues Problem?   Alejandro Kauffmann
• [Asterisk-Users] Re: Asterisk RealTime Voicemail Not Working   Michael Stearne
• [Asterisk-Users] Ring but now audio on answer   Garth Brown
• [Asterisk-Users] Teliax is DOWN   Chris Coulthurst
• [Asterisk-Users] Teliax is DOWN   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Asterisk 1.0.7 on Gentoo   Waldo Rubinstein
• [Asterisk-Users] Teliax is DOWN   Chris Coulthurst
• [Asterisk-Users] Asterisk 1.0.7 on Gentoo   Dan Perik
• [Asterisk-Users] Newbie MP3Player() cmd questions   Henry Junior
• [Asterisk-Users] Pricing for DS3000P   Jason Walker
• [Asterisk-Users] Teliax is DOWN   Chris Coulthurst
• [Asterisk-Users] Teliax is DOWN   Peter Wemm
• [Asterisk-Users] Teliax is DOWN   Chris Mason (Lists)
• [Asterisk-Users] VoiPSupply Dot Com   Wilson Pickett
• [Asterisk-Users] MP3Player could not play remote stream   Phuong Nguyen
• [Asterisk-Users] Teliax is DOWN   Chris Mason (Lists)
• [Asterisk-Users] voip provider request   Jay Milk
• [Asterisk-Users] voip provider request   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Newbie Question: HOWTO make outgoing call on SIP account from internal extensions?   Wilson Pickett
• [Asterisk-Users] Extension context question   Keith Caldwell
• [Asterisk-Users] chan_capi error2   Armin Schindler
• [Asterisk-Users] Connecting Asterisk with Microsoft LCS (Live Communication Server)   Adam Rybak
• [Asterisk-Users] ISDN Data Calls stop working ?   Daniel Nyström
• [Asterisk-Users] voip provider request   Luki
• [Asterisk-Users] Followup: MP3Player cmd issue (for Asterisk OS X users)   Henry Junior
• [Asterisk-Users] [OT] The Voice of Asterisk   Wai-Sun Chia
• [Asterisk-Users] Errors Compiling chan_capi 0.3.5   Armin Schindler
• [Asterisk-Users] Sip UA behind NAT   Eric Yu-Wei Sung
• [Asterisk-Users] Asterisk 1.0.7 on Gentoo   Ron Wellsted
• [Asterisk-Users] voip provider request   Jay Milk
• [Asterisk-Users] Errors Compiling chan_capi 0.3.5   Ralf Schlatterbeck
• [Asterisk-Users] Sip UA behind NAT   Forrest W. Christian
• [Asterisk-Users] Portable USB headset for VoIP   Forrest W. Christian
• [Asterisk-Users] Errors Compiling chan_capi 0.3.5   Armin Schindler
• [Asterisk-Users] Inactivity restart segmentation fault   Alphonse Ogulla
• [Asterisk-Users] Asterisk Realtime - How to enable the debug message for SIP users query   Asterisk User
• [Asterisk-Users] Portable USB headset for VoIP   Mark Benson
• [Asterisk-Users] 911 context, is this right?   Chris Coulthurst
• [Asterisk-Users] voip provider request   trixter http://www.0xdecafbad.com
• [Asterisk-Users] a simple call to my girlfriend   Administrator TOOTAI
• [Asterisk-Users] voip provider request   trixter http://www.0xdecafbad.com
• [Asterisk-Users] secretary function   Christian Hiller
• [Asterisk-Users] * found in Iraq!!   Mark Phillips
• [Asterisk-Users] CRM integration (was RE: CallerID)   Anton Krall
• [Asterisk-Users] a simple call to my girlfriend   Anton Krall
• [Asterisk-Users] How to ensure that software echo cancellation ison?   Anton Krall
• [Asterisk-Users] 4 port BRI options ?   Brett, Gary
• [Asterisk-Users] PAP2-NA with Panasonic KX-TD1232 CE   Zlatko Mesaros
• [Asterisk-Users] 4 port BRI options ?   Nardis Dome
• [Asterisk-Users] How to use same h323-conf-id in incoming and outgoing legs?   Papadopoulos Georgios
• [Asterisk-Users] 4 port BRI options ?   altus
• [Asterisk-Users] 4 port BRI options ?   Brett, Gary
• [Asterisk-Users] Asterisk 1.0.7 on Gentoo   Waldo Rubinstein
• [Asterisk-Users] Any ideas on an Interactive IVR?   cveazey at blackhillsfiber.com
• [Asterisk-Users] Installation of Asterisk addons 1.0.7 fails (longish)   Gundemarie Scholz
• [Asterisk-Users] Teliax is DOWN   Rich Adamson
• [Asterisk-Users] Any ideas on an Interactive IVR?   Time Bandit
• [Asterisk-Users] Asterisk 1.0.7 on Gentoo   Michael George
• [Asterisk-Users] Teliax is DOWN   Chris Mason (Lists)
• [Asterisk-Users] Ring but now audio on answer   Mark Johnson
• [Asterisk-Users] Re: Asterisk 1.0.7 on Gentoo   Maron Kristófersson
• [Asterisk-Users] Re: Asterisk 1.0.7 on Gentoo   Waldo Rubinstein
• [Asterisk-Users] Asterisk 1.0.7 on Gentoo   Waldo Rubinstein
• [Asterisk-Users] Teliax is DOWN   Chris Coulthurst
• [Asterisk-Users] Asterisk 1.0.7 on Gentoo   Waldo Rubinstein
• [Asterisk-Users] SIP_CODEC, reinvites, and changing codecs   Michael George
• [Asterisk-Users] Re: Recommended Network Latency   Maron Kristófersson
• [Asterisk-Users] Asterisk 1.0.7 on Gentoo   Rich Adamson
• [Asterisk-Users] asterisk sipura and g726 codec   Rich Adamson
• [Asterisk-Users] Portable USB headset for VoIP   Mark Benson
• [Asterisk-Users] 4 port BRI options ?   Nardis Dome
• [Asterisk-Users] Asterisk 1.0.7 on Gentoo   Waldo Rubinstein
• [Asterisk-Users] Simple sip.conf question   Matt Schulte
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Rich Adamson
• [Asterisk-Users] 4 port BRI options ?   Gavin Hamill
• [Asterisk-Users] Digium TDM400 Trouble Shooting Tip   Andrew Latham
• [Asterisk-Users] secretary function   Christian Stredicke
• [Asterisk-Users] Re: CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Tony Mountifield
• [Asterisk-Users] G.729 with RVA   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] Asterisk Realtime - How to enable the debug message for SIP users query   Michael Stearne
• [Asterisk-Users] secretary function   Mike Holloway
• [Asterisk-Users] Asterisk @ Home 1.1 Released   Max W Blackmer Jr
• [Asterisk-Users] secretary function   Julian J. M.
• [Asterisk-Users] Everyone-- the scoop on Bison/Flex --   Steve Murphy
• [Asterisk-Users] 4 port BRI options ?   Mark Elkins
• [Asterisk-Users] Sip UA behind NAT   Mike Holloway
• [Asterisk-Users] Teliax is DOWN   Andrew Kohlsmith
• [Asterisk-Users] 4 port BRI options ?   Remco Barende
• [Asterisk-Users] Anybody knows how to setup chan_misdn incoming calls   Rus V. Brushkoff
• [Asterisk-Users] 911 context, is this right?   Rich Adamson
• [Asterisk-Users] Maximum retries exceeded   Tim P
• [Asterisk-Users] 4 port BRI options ?   Remco Barende
• [Asterisk-Users] 4 port BRI options ?   Gavin Hamill
• [Asterisk-Users] * found in Iraq!!   Dean Collins
• [Asterisk-Users] Simple sip.conf question   Rich Adamson
• [Asterisk-Users] G.729 with RVA   Soner Tari
• [Asterisk-Users] Re: CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Rich Adamson
• [Asterisk-Users] voip provider request   Jay Milk
• [Asterisk-Users] 911 context, is this right?   Jay Milk
• [Asterisk-Users] 4 port BRI options ?   Peter Svensson
• [Asterisk-Users] G.729 with RVA   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] 4 port BRI options ?   Florian Overkamp
• [Asterisk-Users] Re: Everyone-- the scoop on Bison/Flex --   Tony Mountifield
• [Asterisk-Users] SNOM 360 extension lights   Ross Kevlin
• [Asterisk-Users] voip provider request   trixter http://www.0xdecafbad.com
• [Asterisk-Users] G.729 with RVA   Soner Tari
• [Asterisk-Users] * found in Iraq!!   C F
• [Asterisk-Users] secretary function   Robert Goodyear
• [Asterisk-Users] Portable USB headset for VoIP   Mark Benson
• [Asterisk-Users] Which version for VAD support?   Eduardo Kaftanski
• [Asterisk-Users] Call Meeting VS Call Confrence   Sandeep A.S
• [Asterisk-Users] oh-323 / Cisco AS5300 problem   Matias G.
• [Asterisk-Users] Ring but now audio on answer   Garth Brown
• [Asterisk-Users] voicemail errors   Joseph
• [Asterisk-Users] chan_sip notices   Joseph
• [Asterisk-Users] Grandstream ATA286 outgoing voice   Erick Perez
• [Asterisk-Users] ring requested on channel 0/23 already in use on span   Robert Goodyear
• [Asterisk-Users] How to ensure that software echo cancellation ison?   Don Murray
• [Asterisk-Users] .call files in outgoing dont get run   Daniel Eriksson
• [Asterisk-Users] Maximum retries exceeded   Rich Adamson
• [Asterisk-Users] SER and Asterisk   Ronald Wiplinger
• [Asterisk-Users] SNOM 360 extension lights   Matias G.
• [Asterisk-Users] Zaptel not found error during modprobe   Richard Lyman
• [Asterisk-Users] New astGUIclient version released 1.1.1   mattf
• [Asterisk-Users] Zaptel not found error during modprobe   Andrew Latham
• [Asterisk-Users] voip provider request   Steve Kennedy
• [Asterisk-Users] Dazed and confused on refresh   Jeffrey Starin
• [Asterisk-Users] How to use same h323-conf-id in incoming and outgoing legs?   Michael Manousos
• [Asterisk-Users] chan_sip notices   Olle E. Johansson
• [Asterisk-Users] Last astcc/* versions working?   C W Nel
• [Asterisk-Users] SNOM 360 extension lights   David John Walsh
• [Asterisk-Users] VoiPSupply Dot Com   Martin List-Petersen
• [Asterisk-Users] a simple call to my girlfriend   Tzafrir Cohen
• [Asterisk-Users] a simple call to my girlfriend   Conrad Wood
• [Asterisk-Users] chan_sip notices   Joseph
• [Asterisk-Users] secretary function   The VoIP Connection
• [Asterisk-Users] Call parking on Polycom 500 doesn't transfer, stays on hold   Chris Coulthurst
• [Asterisk-Users] Re: Everyone-- the scoop on Bison/Flex --   Steve Murphy
• [Asterisk-Users] oh-323 / Cisco AS5300 problem   Leandro Tenorio
• [Asterisk-Users] Call parking on Polycom 500 doesn't transfer, stays on hold   Wiley Siler
• [Asterisk-Users] AgentLogin already on?   Tyler
• [Asterisk-Users] Asterisk - >SIP -> DNIS   Asterisk VoIP
• [Asterisk-Users] a simple call to my girlfriend   Mojo with Horan & Company, LLC
• [Asterisk-Users] oh-323 / Cisco AS5300 problem   Matias G.
• [Asterisk-Users] AgentLogin already on?   Asterisk
• [Asterisk-Users] Call parking on Polycom 500 doesn't transfer, stays on hold   Chris Coulthurst
• [Asterisk-Users] chan_sip notices   Olle E. Johansson
• [Asterisk-Users] add/remove PRI card without rebooting   Patrick Fortin
• [Asterisk-Users] Setting up calls through the manager interface   Bjorn
• [Asterisk-Users] asterisk sipura and g726 codec   Philip Fleischer
• [Asterisk-Users] Can an open source project get acquired?   Dean Collins
• [Asterisk-Users] Call parking on Polycom 500 doesn't transfer, stays on hold   Matthew Marlowe
• [Asterisk-Users] Realtime+IAX2 and RSA   kleis-asterisk-dev at tiscali.it
• [Asterisk-Users] Re: Everyone-- the scoop on Bison/Flex --   Moises Silva
• [Asterisk-Users] Re: Announce: Asterisk virtual configuration   snacktime
• [Asterisk-Users] SNOM 360 extension lights   Ross Kevlin
• [Asterisk-Users] oh-323 / Cisco AS5300 problem   Leandro Tenorio
• [Asterisk-Users] Karl   Karl J. Vesterling
• [Asterisk-Users] Karl   Karl J. Vesterling
• [Asterisk-Users] Help Cisco 12+   Guruirl
• [Asterisk-Users] SNOM 360 extension lights   Olle E. Johansson
• [Asterisk-Users] Use with Octtel 8 port fxs device   Guruirl
• [Asterisk-Users] Call parking on Polycom 500 doesn't transfer, stays on hold   Chris Coulthurst
• [Asterisk-Users] Call Routing based on number dialed (using SIP)   Geoff Manning
• [Asterisk-Users] Interesting article on new SIP phones   Michael Puchol
• [Asterisk-Users] SNOM 360 extension lights   Ross Kevlin
• [Asterisk-Users] VoiPSupply Dot Com   Karl J. Vesterling
• [Asterisk-Users] oh-323 / Cisco AS5300 problem   Pavel Jezek
• E911 Legislation (was) Re: [Asterisk-Users] VoiPSupply Dot Com   Karl J. Vesterling
• [Asterisk-Users] login/logout of call queue   Scott
• [Asterisk-Users] Livevoip 800 Choppy Audio   John Kington
• [Asterisk-Users] Martin Zimmermann is not in the office   Martin.Zimmermann at db-telematik.de
• [Asterisk-Users] Martin Zimmermann is not in the office   Dean Collins
• [Asterisk-Users] Livevoip 800 Choppy Audio   Wiley Siler
• [Asterisk-Users] a simple call to my girlfriend   Filippo Carone
• [Asterisk-Users] Martin Zimmermann is not in the office   Filippo Carone
• [Asterisk-Users] Livevoip 800 Choppy Audio   Scott Wolfe
• [Asterisk-Users] Livevoip 800 Choppy Audio   Wiley Siler
• [Asterisk-Users] (no subject)   support at sjobeck.com
• [Asterisk-Users] Problem starting RX_FAX and TX_FAX Module   Jorge Alayon
• [Asterisk-Users] Asterisk and Audiocodes 108 FXS   Jorge Alayon
• [Asterisk-Users] Realtime+IAX2 and RSA   pbx at itsngroup.com
• [Asterisk-Users] login/logout of call queue   Adam Vocks
• [Asterisk-Users] asterisk sipura and g726 codec   Luki
• [Asterisk-Users] voip provider request   Stanley Cline
• [Asterisk-Users] Incoming and Outgoing   Andy Hamilton
• [Asterisk-Users] spam filter   Amicus Curious
• [Asterisk-Users] DID Routing over SIP   Andy Hamilton
• [Asterisk-Users] How to ensure that software echo cancellation ison?   Don Murray
• [Asterisk-Users] How to ensure that software echo cancellation ison?   Don Murray
• [Asterisk-Users] Can an open source project get acquired?   Andrew Latham
• [Asterisk-Users] Caller ID Routing using VoicePulseConnect   pbx at itsngroup.com
• [Asterisk-Users] Martin Zimmermann is not in the office   Jay Milk
• [Asterisk-Users] Problem starting RX_FAX and TX_FAX Module   Steve Underwood
• [Asterisk-Users] ARESKICC DOESN'T make a CALL!!!   Robson Ribeiro
• [Asterisk-Users] ztdummy errors on WBEL4   geek at mnwifi.net
• [Asterisk-Users] Pricing for DS3000P   Matt Klein
• [Asterisk-Users] Pricing for DS3000P   Matt Klein
• [Asterisk-Users] How to ensure that software echo cancellation ison?   Rich Adamson
• [Asterisk-Users] Zap channel not hangingup   rajkumars at asianetindia.com
• [Asterisk-Users] Errors Compiling chan_capi 0.3.5   Ralf Schlatterbeck
• [Asterisk-Users] SNOM 360 extension lights   Olle E. Johansson
• [Asterisk-Users] Re: ztdummy errors on WBEL4   Tony Mountifield
• SV: [Asterisk-Users] Setting up calls through the manager interface   Bjorn
• [Asterisk-Users] Re: Segfault on restart   Alphonse Ogulla
• [Asterisk-Users] Pricing for DS3000P   Tom Fanning
• [Asterisk-Users] SNOM extension lights programmable, eg. based on asterisk variable setting?   Bruno.Voigt at ic3s.de
• SV: [Asterisk-Users] Setting up calls through the manager interface   Olle E. Johansson
• [Asterisk-Users] Pricing for DS3000P   Pavel Jezek
• [Asterisk-Users] Is there any SCCP patent issue?   Pavel Jezek
• [Asterisk-Users] Pricing for DS3000P   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Pricing for DS3000P   Peter Svensson
• [Asterisk-Users] SMS + SMSQ   Henry Junior
• E911 Legislation (was) Re: [Asterisk-Users] VoiPSupply Dot Com   Chris Travers
• [Asterisk-Users] SNOM extension lights programmable, eg. based on asterisk variable setting?   Olle E. Johansson
• [Asterisk-Users] Re: Setting up calls through the manager interface   Tony Mountifield
• [Asterisk-Users] Automatic callback feature *66   Chris Coulthurst
• [Asterisk-Users] Caller ID Routing using VoicePulseConnect   Chris Coulthurst
• [Asterisk-Users] a simple call to my girlfriend   Hendrik Wouters
• [Asterisk-Users] [Fwd: Help Cisco 12+]   Guruirl
• [Asterisk-Users] [Fwd: Use with Octtel 8 port fxs device]   Guruirl
• [Asterisk-Users] [Fwd: [Fwd: Use with Octtel 8 port fxs device]]   Guruirl
• [Asterisk-Users] [Fwd: [Fwd: Help Cisco 12+]]   Guruirl
• [Asterisk-Users] Incoming SIP calls with no extension   Christoph Weber
• [Asterisk-Users] Re: [Asterisk-Dev] Is there any SCCP patent issue?   Julien Goodwin
• [Asterisk-Users] Incoming SIP calls with no extension   Torsten Krueger
• [Asterisk-Users] Pricing for DS3000P   Steve Underwood
• [Asterisk-Users] chan_sccp / 7960: Messages key, line / speeddial keys   Julien Goodwin
• [Asterisk-Users] Pricing for DS3000P   Matthew Crocker
• [Asterisk-Users] Apologies   Guruirl
• [Asterisk-Users] call queues problem   Sander
• [Asterisk-Users] Pricing for DS3000P   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Errors Compiling chan_capi 0.3.5   Armin Schindler
• [Asterisk-Users] Pricing for DS3000P   Matt Klein
• [Asterisk-Users] Problem with X100P (ZT_SPANCONFIG failed)   Dakapo
• [Asterisk-Users] chan_sip + MD5 encryption: WARNING Format for authentication entry is user[:secret]@realm   kleis-asterisk-dev at tiscali.it
• [Asterisk-Users] Pricing for DS3000P   Matt Klein
• [Asterisk-Users] call queues problem   Guido Hecken
• [Asterisk-Users] Pricing for DS3000P   Matt Klein
• [Asterisk-Users] Pricing for DS3000P   Matt Klein
• [Asterisk-Users] .call files in outgoing dont get run   Chris Travers
• [Asterisk-Users] chan_sip + MD5 encryption: WARNING Format for authentication entry is user[:secret]@realm   Olle E. Johansson
• RE: [Asterisk-Users] chan_sip + MD5 encryption: WARNING Format for authentication entry is user[:secret]@realm   kleis-asterisk-dev at tiscali.it
• [Asterisk-Users] AgentCallbackLogin   Steven Lam
• [Asterisk-Users] AgentCallbackLogin   Steven Lam
• [Asterisk-Users] Errors Compiling chan_capi 0.3.5   Ralf Schlatterbeck
• [Asterisk-Users] Errors Compiling chan_capi 0.3.5   Armin Schindler
• [Asterisk-Users] AgentCallbackLogin   Kevin Bockman
• [Asterisk-Users] Possibly a bug? (Was: Setting up calls through the manager interface)   Bjorn
• [Asterisk-Users] Errors Compiling chan_capi 0.3.5   Ralf Schlatterbeck
• [Asterisk-Users] Pricing for DS3000P   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Automatic callback feature *66   Jay Milk
• [Asterisk-Users] Pricing for DS3000P   Jay Milk
• [Asterisk-Users] Pricing for DS3000P   Michael D Schelin
• [Asterisk-Users] AgentCallbackLogin   Steven Lam
• SV: [Asterisk-Users] Possibly a bug? (Was: Setting up calls through themanager interface)   Bjorn
• [Asterisk-Users] Automatic callback feature *66   Chris Coulthurst
• [Asterisk-Users] Pricing for DS3000P   Andrew Latham
• [Asterisk-Users] UNSUBSCRIBE   Jake Rog
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Steve
• [Asterisk-Users] Pricing for DS3000P   trixter http://www.0xdecafbad.com
• [Asterisk-Users] How to ensure that software echo cancellation ison?   Michael D Schelin
• [Asterisk-Users] Automatic callback feature *66   Chris Coulthurst
• [Asterisk-Users] How to quickly replace ',' with '|' in dialplans?   kleis-asterisk-dev at tiscali.it
• [Asterisk-Users] How to quickly replace ', ' with '|' in dialplans?   Steve
• [Asterisk-Users] X100P installed OK, after added TDM400P Asterisk would no longer start   Angus Comber
• [Asterisk-Users] Errors Compiling chan_capi 0.3.5   Armin Schindler
• [Asterisk-Users] How to quickly replace ', ' with '|' in dialplans?   Michael D Schelin
• [Asterisk-Users] Garbled speech - strange problem.   Chris Mason (Lists)
• [Asterisk-Users] Errors Compiling chan_capi 0.3.5   Ralf Schlatterbeck
• [Asterisk-Users] X100P installed OK, after added TDM400P Asterisk would no longer start   Ralf Schlatterbeck
• Re: [Asterisk-Users] How to quickly replace ', ' with '|' in dialplans?   kleis-asterisk-dev at tiscali.it
• Re: [Asterisk-Users] How to quickly replace ', ' with '|' in dialplans?   kleis-asterisk-dev at tiscali.it
• [Asterisk-Users] Asterisk@Home Forum Suggestions   Tracy Phillips
• [Asterisk-Users] Pricing for DS3000P   Matt Klein
• [Asterisk-Users] Pricing for DS3000P   Matt Klein
• [Asterisk-Users] Pricing for DS3000P   Matt Klein
• [Asterisk-Users] facing problems with TDM400P   wassim darwish
• [Asterisk-Users] Satelite Internet connection   Kumara Jayaweera
• Re: [Asterisk-Users] chan_sip + MD5 encryption: WARNING Format for authentication entry is user[:secret]@realm   kleis-asterisk-dev at tiscali.it
• [Asterisk-Users] Pricing for DS3000P   Andrew Kohlsmith
• [Asterisk-Users] Pricing for DS3000P   Andrew Kohlsmith
• [Asterisk-Users] X100P installed OK, after added TDM400P Asterisk would no longer start   Angus Comber
• [Asterisk-Users] VoiPSupply Dot Com   Andrew Kohlsmith
• [Asterisk-Users] Pricing for DS3000P   Andrew Kohlsmith
• [Asterisk-Users] Pricing for DS3000P   Forrest W. Christian
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Steve
• [Asterisk-Users] Pricing for DS3000P   Matt Klein
• [Asterisk-Users] Pricing for DS3000P   Matt Klein
• [Asterisk-Users] Pricing for DS3000P   Matt Klein
• [Asterisk-Users] How to quickly replace ', ' with '|' in dialplans?   Iqbal
• [Asterisk-Users] How to quickly replace ', ' with '|' in dialplans?   Jeremy McDermond
• [Asterisk-Users] How to quickly replace ', ' with '|' in dialplans?   Steve
• [Asterisk-Users] X100P installed OK, after added TDM400P Asterisk would no longer start   Ralf Schlatterbeck
• [Asterisk-Users] Pricing for DS3000P   Andrew Latham
• [Asterisk-Users] How to quickly replace ', ' with '|' in dialplans?   Chris Coulthurst
• [Asterisk-Users] zap to zap bridging not hanging up   Master Abi
• [Asterisk-Users] SetCallerID based on extension   Jodie Crouch
• [Asterisk-Users] Extension 'hint' info please?   Chris Coulthurst
• [Asterisk-Users] Extension 'hint' info please?   Ed Greenberg
• [Asterisk-Users] zap to zap bridging not hanging up   Paradise Dove
• [Asterisk-Users] SetCallerID based on extension   Chris Coulthurst
• [Asterisk-Users] New version of IPSwitchBoard   Thorben Jensen
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Olle E. Johansson
• [Asterisk-Users] Unable to create channel of type SIP-please help   asterisk
• [Asterisk-Users] Extension 'hint' info please?   Olle E. Johansson
• [Asterisk-Users] Cisco Softphone 1.3(4a) issue.   Mohamed A. Gombolaty
• [Asterisk-Users] New version of Asterisk VConfig   snacktime
• [Asterisk-Users] CDR records.. How do you deal with them?   Bjorn
• [Asterisk-Users] Compilation on Debian with support for HFC-chip based ISDN-cards   Klaus Ruebsam
• [Asterisk-Users] zap to zap bridging not hanging up   Rajkumar S
• [Asterisk-Users] Pricing for DS3000P   Jean-Michel Hiver
• [Asterisk-Users] zap to zap bridging not hanging up   Andrew Kohlsmith
• [Asterisk-Users] Compilation on Debian with support for HFC-chip based ISDN-cards   Martin List-Petersen
• [Asterisk-Users] Pricing for DS3000P   Andrew Kohlsmith
• [Asterisk-Users] Pricing for DS3000P   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Pricing for DS3000P   Andrew Kohlsmith
• [Asterisk-Users] Asterisk@Home Forum Suggestions   Dean Collins
• [Asterisk-Users] Pricing for DS3000P   trixter http://www.0xdecafbad.com
• [Asterisk-Users] 180 Ringing?   Mirko Marghitola
• [Asterisk-Users] X100P installed OK, after added TDM400P Asterisk would no longer start   Angus Comber
• [Asterisk-Users] Pricing for DS3000P   Andrew Kohlsmith
• [Asterisk-Users] Pricing for DS3000P   Steve Underwood
• [Asterisk-Users] Pricing for DS3000P   Andrew Kohlsmith
• [Asterisk-Users] Pricing for DS3000P   John Millican
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Chris Mason (Lists)
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Jean-Michel Hiver
• [Asterisk-Users] Pricing for DS3000P   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Gavin Hamill
• [Asterisk-Users] zap to zap bridging not hanging up   Rajkumar S
• [Asterisk-Users] DTMF Tone Lengths   Ronald Hartmann
• [Asterisk-Users] DTMF Tone Lengths   Rich Adamson
• [Asterisk-Users] New version of Asterisk VConfig   snacktime
• [Asterisk-Users] ACD Login   adam.collard at sbcglobal.net
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Steve
• [Asterisk-Users] UK call disconnects during record   Scott England
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Chris Mason (Lists)
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Zoa
• [Asterisk-Users] VoiPSupply Dot Com   C F
• [Asterisk-Users] Karl   C F
• [Asterisk-Users] Outgoing TDM400P FXO calls always answered   Sam Lown
• [Asterisk-Users] VoiPSupply Dot Com   steve szmidt
• [Asterisk-Users] te410p not working after cvs-head update   Asterisk
• [Asterisk-Users] te410p not working after cvs-head update   Kevin Bockman
• [Asterisk-Users] te410p not working after cvs-head update   Asterisk
• [Asterisk-Users] te410p not working after cvs-head update   Asterisk
• [Asterisk-Users] Pricing for DS3000P   Andrew Kohlsmith
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Chris Mason (Lists)
• [Asterisk-Users] ISDN 4 BRI card for UK   Angus Comber
• [Asterisk-Users] IAXtel update!   Russell Bryant
• [Asterisk-Users] Re: ISDN 4 BRI card for UK   Tony Mountifield
• [Asterisk-Users] 180 Ringing?   Olle E. Johansson
• [Asterisk-Users] Re: ISDN 4 BRI card for UK   Martin List-Petersen
• [Asterisk-Users] Re: ISDN 4 BRI card for UK   Tony Mountifield
• [Asterisk-Users] Examples of Asterisk deployments with 100-500 users?   Roman Zhovtulya
• [Asterisk-Users] VoiceMail Termporary greeting option   C F
• [Asterisk-Users] Adtran 600 channel bank   Chris Mason (Lists)
• [Asterisk-Users] Disa - how it returns on user not dialing any numbers ?   Robert Rozman
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Rich Adamson
• [Asterisk-Users] Moh   時期 精霊
• [Asterisk-Users] sipura3000 problems in callcenter   Dan Fernandez
• [Asterisk-Users] Disa - how it returns on user not dialing any numbers ?   C F
• [Asterisk-Users] zap to zap bridging not hanging up   Andrew Kohlsmith
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Kevin P. Fleming
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Kevin P. Fleming
• [Asterisk-Users] Disa - how it returns on user not dialing anynumbers ?   Robert Rozman
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Steve Underwood
• [Asterisk-Users] Voice Dtect   Anton Krall
• [Asterisk-Users] Disa - how it returns on user not dialing anynumbers ?   C F
• [Asterisk-Users] TDM20B FXS card configuration?   Kumara Jayaweera
• [Asterisk-Users] zap to zap bridging not hanging up   Rich Adamson
• [Asterisk-Users] IAXtel update!   Rich Adamson
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Rich Adamson
• [Asterisk-Users] TDM20B FXS card configuration?   Rich Adamson
• [Asterisk-Users] Voice Dtect   Matt Riddell
• [Asterisk-Users] 911 context, is this right?   Adam Goryachev
• [Asterisk-Users] Call parking on Polycom 500 doesn't transfer, stays on hold   Adam Goryachev
• [Asterisk-Users] Accountcode being ignored?   Nabeel Jafferali
• [Asterisk-Users] A Few Questions   C. Hatton Humphrey
• [Asterisk-Users] 911 context, is this right?   Rich Adamson
• [Asterisk-Users] A Few Questions   adam.collard at sbcglobal.net
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Brian McSpadden
• [Asterisk-Users] 911 context, is this right?   Adam Goryachev
• [Asterisk-Users] Answered but No Answer   David Choo
• [Asterisk-Users] Problems getting VoicePulse Connect working   adam.collard at sbcglobal.net
• [Asterisk-Users] zap to zap bridging not hanging up   Rajkumar S
• [Asterisk-Users] Sip UA behind NAT   Eric Yu-Wei Sung
• [Asterisk-Users] TDM400P Polarity reversal detection   Boris Bakchiev
• [Asterisk-Users] RXFax and Hangup context Question.   pbx at itsngroup.com
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Steve
• [Asterisk-Users] A Few Questions   Matt Riddell
• [Asterisk-Users] TDM400P Polarity reversal detection   Matt Riddell
• [Asterisk-Users] TDM400P Polarity reversal detection   Matt Riddell
• [Asterisk-Users] Disa - how it returns on user not dialinganynumbers ?   Robert Rozman
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Steve
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Steve
• [Asterisk-Users] Re: Bison, Flex, Conditional Expression   Steve Murphy
• [Asterisk-Users] handytone with asterisk   Betül Gözlükoğlu
• [Asterisk-Users] Little help with MySQL please   Ronald Wiplinger
• [Asterisk-Users] H323 trunk with cisco gatekeeper   Mahmoud Badran
• [Asterisk-Users] Accountcode being ignored?   Olle E. Johansson
• [Asterisk-Users] SIP changes in CVS head   Olle E. Johansson
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Kib Eki
• [Asterisk-Users] Disa - how it returns on user not dialing any numbers ?   Peter Svensson
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Peter Nixon
• [Asterisk-Users] Cisco 7960s and skinny   Yusuf Iqbal
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Peter Nixon
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Nir Simionovich
• [Asterisk-Users] mISDN + chan_misdn.so + winbond issue   Michel Koenen
• [Asterisk-Users] No DTMF interpretation on outgoing calls   JunkMail
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Gavin Hamill
• [Asterisk-Users] Variables and status problems in AGI application   asterisklists at adontendev.net
• music on hold [was: Re: [Asterisk-Users] A Few Questions]   Tzafrir Cohen
• [Asterisk-Users] Issue with SIP inter-op   Nir Simionovich
• [Asterisk-Users] te410p not working after cvs-head update   Asterisk
• [Asterisk-Users] Quotation request: 12 KHz signal generation for billing purposes.   Cenk Yabas
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Chris Mason (Lists)
• [Asterisk-Users] what hardware components do i need?   infra struct
• [Asterisk-Users] Quotation request: 12 KHz signal generation for billing purposes.   Marc Storck
• [Asterisk-Users] Issue with SIP inter-op   Olle E. Johansson
• [Asterisk-Users] Quotation request: 12 KHz signal generation for billing purposes.   Matt Riddell
• [Asterisk-Users] Compilation on Debian with support for HFC-chip based ISDN-cards   Tzafrir Cohen
• [Asterisk-Users] Compiling asterisk-addons-1.0.7 on Debian Sarge with asterisk-packages installed   Christian Wengel
• [Asterisk-Users] Bluetooth with *   fredrik chabot
• [Asterisk-Users] 911 context, is this right?   Andrew Kohlsmith
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Kib Eki
• [Asterisk-Users] How to Playback a file continuously during conversation?   Papadopoulos Georgios
• [Asterisk-Users] No DTMF interpretation on outgoing calls   JunkMail
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Rich Adamson
• [Asterisk-Users] How to Playback a file continuously during conversation?   Joseph
• [Asterisk-Users] AMP and custom application   pellegrini at frameweb.it
• [Asterisk-Users] BN8S0 crash linux on connect   me me
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Chris Mason (Lists)
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Rich Adamson
• [Asterisk-Users] Moh   時期 精霊
• [Asterisk-Users] 911 context, is this right?   Adam Goryachev
• [SPAM] - [Asterisk-Users] what hardware components do i need? - Email found in subject   Betül Gözlükoğlu
• [Asterisk-Users] snom 360 conference button   Christian Hiller
• [Asterisk-Users] SNOM 360 extension lights   Ross Kevlin
• [Asterisk-Users] Features.conf - atxfer   Mark Johnson
• [Asterisk-Users] Accountcode being ignored?   Aza
• [Asterisk-Users] Echo Issues via SIP   Graham Pearson
• [Asterisk-Users] Asterisk and Audiocodes 108 FXS   Steve Clark
• [Asterisk-Users] IRQ Problems   Tom Hayden
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Tim Pushor
• [Asterisk-Users] Any thoughts   Marcelo Pacheco
• [Asterisk-Users] IRQ Problems   Doug Lytle
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Peter Svensson
• [Asterisk-Users] IRQ Problems   Juan Pablo Abuyeres
• [Asterisk-Users] IRQ Problems   Zoa
• [Asterisk-Users] IRQ Problems   Gustavo Alvarez
• [Asterisk-Users] IRQ Problems   Tom Hayden
• [Asterisk-Users] A Few Questions   adam.collard at sbcglobal.net
• [Asterisk-Users] 911 context, is this right?   Rich Adamson
• [Asterisk-Users] ARESKICC DOESN'T make a CALL!!!   Areski K
• [Asterisk-Users] A Few Questions   C. Hatton Humphrey
• [Asterisk-Users] Zaptel comple on FC2   Mark Benson
• [Asterisk-Users] IRQ Problems   Gustavo Alvarez
• [Asterisk-Users] IRQ Problems   Zoa
• [Asterisk-Users] Answering a queue with an SIP UA, then transfer to another sip UA oneway audio   fredrik chabot
• [Asterisk-Users] Zaptel comple on FC2   Mark Benson
• [Asterisk-Users] Polycom 500...   Carlos Chavez
• [Asterisk-Users] No DTMF interpretation on outgoing calls   Jay Milk
• [Asterisk-Users] SNOM extension lights programmable, eg. based on asterisk variable setting?   Josh Dady
• [Asterisk-Users] A Few Questions   Dean Collins
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Kevin P. Fleming
• [Asterisk-Users] IAX Phone Pro - Open Beta Test   Steven Sokol
• [Asterisk-Users] Variables and status problems in AGI application   asterisklists at adontendev.net
• [Asterisk-Users] Polycom 500...   Mark Benson
• [Asterisk-Users] No DTMF interpretation on outgoing calls   Rich Adamson
• [Asterisk-Users] AMP and custom application   Moises Silva
• [Asterisk-Users] A Few Questions   Mark Benson
• [Asterisk-Users] Asterisk at Home ...   Alan Bunch
• [Asterisk-Users] OT: WAS: * found in Iraq!! NOW: Asterisk bus iness sightings   Colin Anderson
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   William Waites
• [Asterisk-Users] Asterisk at Home ...   barney
• [Asterisk-Users] Variables and status problems in AGI application   Brian C. Fertig
• [Asterisk-Users] IRQ Problems   Rich Adamson
• [Asterisk-Users] How to make Polycom phones work with Asterisk as a SIP Client?   Kanuri, Seshu (Company IT)
• [Asterisk-Users] D channel initialization   Jorge Verastegui
• [Asterisk-Users] Re: 4 port BRI options ?   Louis-David Mitterrand
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Andrew Kohlsmith
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Max W Blackmer Jr
• [Asterisk-Users] IAX Phone Pro - Open Beta Test   Steven Sokol
• R: R: R: [Asterisk-Users] AT-320 + supervised transfer   Giordano Grandis
• [Asterisk-Users] How to make Polycom phones work with Asterisk as aSIP Client?   Wiley Siler
• [Asterisk-Users] IRQ Problems   Tom Hayden
• [Asterisk-Users] Pricing for DS3000P   Chris Modesitt
• [Asterisk-Users] Hangupcode == 44   Tom Hayden
• [Asterisk-Users] OT: Please comment on Dvorak's troll   Colin Anderson
• [Asterisk-Users] IRQ Problems   Doug Lytle
• [Asterisk-Users] IRQ Problems   Craig Guy
• [Asterisk-Users] PHP/AGI Problem   asterisklists at adontendev.net
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   William Waites
• [Asterisk-Users] Pricing for DS3000P   Andrew Kohlsmith
• [Asterisk-Users] IRQ Problems   Tom Hayden
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Brian Capouch
• [Asterisk-Users] G.729 with RVA   Nathaniel Angelo A. Torres (247talk)
• [Asterisk-Users] How to make Polycom phones work with Asterisk asaSIP Client?   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Dial(SIP/xyz&zap/r1/123) with different Caller Ids ?!   Bruno.Voigt at ic3s.de
• [Asterisk-Users] Degraded voice without packet loss   MF Hulber
• [Asterisk-Users] Asterisk Live! CF   abel
• [Asterisk-Users] Asterisk Live! CF   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Unable to Configure NetPhone IP phone   SYED ADEEL ALI
• [Asterisk-Users] Asterisk Live! CF   abel
• [Asterisk-Users] Newbee, help with cdr/odbc/mysql logging problem   monty-asterisk at bc.cx
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Steve
• [Asterisk-Users] Re: Polycom 500...   Noah Miller
• [Asterisk-Users] Re: ISDN 4 BRI card for UK   Leandro Morgado
• [Asterisk-Users] Cisco 7960 + 7914 using SIP   Oswaldo Arratia
• [Asterisk-Users] Re: Polycom 500...   Don Murray
• [Asterisk-Users] Quotation request: 12 KHz signal generation for billing purposes.   izo
• [Asterisk-Users] Re: Polycom 500...   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Re: CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Tony Mountifield
• [Asterisk-Users] Incoming Calls   David Sampson
• [Asterisk-Users] Cisco 7960 + 7914 using SIP   Joseph
• [Asterisk-Users] Cisco 7960 + 7914 using SIP   Robert Lawrence
• [Asterisk-Users] Erro message - Received mini frame before first full voice frame   Chris Mason (Lists)
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Andrew Kohlsmith
• [Asterisk-Users] Re: Polycom 500...   Noah Miller
• [Asterisk-Users] OT: Please comment on Dvorak's troll   Dan Littlejohn
• [Asterisk-Users] Asterisk Live! CF   Kristian Kielhofner
• [Asterisk-Users] dial-a-string (e.g. 2=a,b,c, 3=d,e,f)   Ivan Fetch
• [Asterisk-Users] A Few Questions   Matt
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outboundsip call to PSTN   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Cisco 7960 + 7914 using SIP   Oswaldo Arratia
• R: R: R: [Asterisk-Users] AT-320 + supervised transfer   Gavin Hamill
• [Asterisk-Users] Re: ISDN 4 BRI card for UK   Emanuele Pucciarelli
• [Asterisk-Users] Re: 4 port BRI options ?   Armin Schindler
• [Asterisk-Users] whining on the list   Jon Pounder
• [Asterisk-Users] Re: SNOM extension lights programmable, eg. based on asterisk variable setting?   Bruno.Voigt at ic3s.de
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Rich Adamson
• [Asterisk-Users] Asterisk Live! CF   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Asterisk eating up 99.8% cpu   Umair Bari
• [Asterisk-Users] Re: CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Steve
• [Asterisk-Users] OT: Please comment on Dvorak's troll   Colin Anderson
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   William Waites
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outboundsip call to PSTN   Steve
• [Asterisk-Users] *@home .conf files request   Luis Diaz
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   William Waites
• [Asterisk-Users] Double NAT issues with SIP and workaround (?)   Julian J. M.
• [Asterisk-Users] Incoming Calls   Carlos Chavez
• [Asterisk-Users] Compiling asterisk-addons-1.0.7 on Debian Sarge with asterisk-packages installed   Tzafrir Cohen
• [Asterisk-Users] Servers Compatible with Digium HW   Syed Akbar
• [Asterisk-Users] ENUM NL dead ?   Michiel van Baak
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outboundsip call to PSTN   Steve
• [Asterisk-Users] A Few Questions   Tzafrir Cohen
• [Asterisk-Users] Double NAT issues with SIP and workaround (?)   Steve
• [Asterisk-Users] CLUELESS NEWBIE needs help making an outboundsip call to PSTN   Steve
• [Asterisk-Users] ADSI over SIP   John Bittner
• [Asterisk-Users] mISDN + chan_misdn.so + winbond issue   David Phelan
• [Asterisk-Users] Transfer differences between BudgeTone101 and Snom190   Elwin Andriol
• [Asterisk-Users] About BillSec when having conferences   Moises Silva
• [Asterisk-Users] Polycom 500...   C F
• [Asterisk-Users] Transfer differences between BudgeTone101 and Snom190   Greg Oliver
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Andrew Kohlsmith
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Chris Mason (Lists)
• [Asterisk-Users] OT: Please comment on Dvorak's troll   Michael Stearne
• [Asterisk-Users] *@home .conf files request   Dan Perik
• [Asterisk-Users] Erro message - Received mini frame before first full voice frame   Andrew Kohlsmith
• [Asterisk-Users] *@home .conf files request   Terry H. Gilsenan
• [Asterisk-Users] NAT & RealTime   Jeromy Grimmett
• [Asterisk-Users] Debugging SIP Connection   Graham Pearson
• [Asterisk-Users] Last astcc/* versions working?   Darren Wiebe
• [Asterisk-Users] Problems getting VoicePulse Connect working   Tracy Phillips
• [Asterisk-Users] OT: Please comment on Dvorak's troll   Dean Collins
• [Asterisk-Users] Re: CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Steve
• [Asterisk-Users] Service Unavailble, Sipura 3000, CheckGroup, what the heck??   Dan Fernandez
• [Asterisk-Users] Re: Polycom 500...   rsenykoff at harrislogic.com
• [Asterisk-Users] Asterisk Live! CF   Kristian Kielhofner
• [Asterisk-Users] Issue with IAXy in Canada?   Obaid Siddiqui
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Andrew Latham
• [Asterisk-Users] Issue with IAXy in Canada?   Jon Pounder
• [Asterisk-Users] Accountcode being ignored?   Nabeel Jafferali
• [Asterisk-Users] Accountcode being ignored?   Nabeel Jafferali
• [Asterisk-Users] Issue with IAXy in Canada?   Andrew Kohlsmith
• [Asterisk-Users] Service Unavailble, Sipura 3000, CheckGroup, what the heck??   Rich Adamson
• [Asterisk-Users] Issue with IAXy in Canada?   Obaid Siddiqui
• [Asterisk-Users] RE: LOA for CFA . . work up "pencil copy"   Shaun Tierney
• [Asterisk-Users] RE: LOA for CFA . . work up "pencil copy"   brett at websmyths.com
• [Asterisk-Users] Disa - how it returns on user not dialing anynumbers ?   Robert Rozman
• [Asterisk-Users] Debugging SIP Connection   Ritesh Jalan
• [Asterisk-Users] Servers Compatible with Digium HW   Jason Walker
• [Asterisk-Users] ENUM NL dead ?   Florian Overkamp
• [Asterisk-Users] Karl   Ronan Eckelberry
• [Asterisk-Users] Re: chan_sccp / 7960: "External call" and more   Sergio Chersovani
• [Asterisk-Users] Servers Compatible with Digium HW   Syed Akbar
• [Asterisk-Users] problem accepting sip call cvs head   rchen
• [Asterisk-Users] Digium G729 licensing - is it worth the trouble?   Robert Goodyear
• [Asterisk-Users] USB phones...   Gary
• [Asterisk-Users] run a script on completion of call   Eric Smith - Fruitcom
• [Asterisk-Users] How to allow multiple codecs in A@H   Erdem HAKİ
• [Asterisk-Users] ENUM NL dead ?   Michiel van Baak
• [Asterisk-Users] Problems with Junghanns QuadBri   Daniele Gianetti
• [Asterisk-Users] AMP and custom application   pellegrini at frameweb.it
• [Asterisk-Users] Duplicate Calls   VoIP Newbie
• [Asterisk-Users] ENUM NL dead ?   Francesco Peeters
• R: R: R: R: [Asterisk-Users] AT-320 + supervised transfer   Giordano Grandis
• [Asterisk-Users] Problems with Junghanns QuadBri   Daniele
• [Asterisk-Users] connecting Asterisk with Siemens HiPath4000   Ohad.Levy at infineon.com
• [Asterisk-Users] Problems with Junghanns QuadBri   Dpto. Técnico.
• [Asterisk-Users] Re: ISDN 4 BRI card for UK   Amatisoft SRL
• [Asterisk-Users] te405p and dell poweredge   Ma Zhiyong
• [Asterisk-Users] D-link DPH-80 (SIP) call to asterisk problem   Eugene Crosser
• [Asterisk-Users] OT: Please comment on Dvorak's troll   MF Hulber
• [Asterisk-Users] How to configure 2 asterisks   Cleyverson P. Costa
• [Asterisk-Users] I want to move the MySQL server out to another machine   Ronald Wiplinger
• [Asterisk-Users] I want to move the MySQL server out to anothermachine   Ritesh Jalan
• [Asterisk-Users] Sip UA behind NAT   Mike Holloway
• [Asterisk-Users] I want to move the MySQL server out to another machine   asterisklists at adontendev.net
• [Asterisk-Users] I want to move the MySQL server out to anothermachine   Mike Holloway
• [Asterisk-Users] what hardware components do i need?   Mike Holloway
• [Asterisk-Users] 180 Ringing? (BUG?)   Michael Manousos
• [Asterisk-Users] Re: 4 port BRI options ?   Gavin Hamill
• [Asterisk-Users] Features.conf - atxfer   Mike Holloway
• [Asterisk-Users] English vs American voice files   Mark Phillips
• [Asterisk-Users] RE: Asterisk-Users] te405p and dell poweredge   Muhammad Nasim
• [Asterisk-Users] I want to move the MySQL server out to anothermachine   Ronald Wiplinger
• [Asterisk-Users] Features.conf - atxfer   Greg Oliver
• [Asterisk-Users] I want to move the MySQL server out to anothermachine   Ritesh Jalan
• [Asterisk-Users] I want to move the MySQL server out to anothermachine   Mike Holloway
• [Asterisk-Users] OT: Please comment on Dvorak's troll   Michael Graves
• [Asterisk-Users] Call Routing based on number dialed (using S IP)   Geoff Manning
• [Asterisk-Users] USB phones...   Matt Riddell
• [Asterisk-Users] ENUM NL dead ?   Asterisk
• [Asterisk-Users] I want to move the MySQL server out to anothermachine   Mike Holloway
• [Asterisk-Users] connecting Asterisk with Siemens HiPath4000   richard Coco
• [Asterisk-Users] RE: Asterisk-Users] te405p and dell poweredge   Aza
• [Asterisk-Users] Re: Answering a queue with an SIP UA, then transfer to another sip UA oneway audio   fredrik chabot
• [Asterisk-Users] NEWBIE: sip subscriptions   Ciprian Cosma
• [Asterisk-Users] Issue with IAXy in Canada?   Rich Adamson
• [Asterisk-Users] CallerID/chan_sccp   Joseph
• [Asterisk-Users] Call Routing based on number dialed (using S IP)   Mirko Marghitola
• [Asterisk-Users] 3com 3105 Attendant DSS Console (SIP??)   Chris Coulthurst
• [Asterisk-Users] NAT & RealTime   Matthew Boehm
• [Asterisk-Users] Issue with IAXy in Canada?   Robert Webb
• [Asterisk-Users] run a script on completion of call   Chris Coulthurst
• [Asterisk-Users] Features.conf - atxfer   Dennis Walker
• [Asterisk-Users] te405p and dell poweredge   Matthew Boehm
• [Asterisk-Users] (no subject)   Mark Ackroyd
• [Asterisk-Users] gxp-2000 tftp cfg   marek cervenka
• [Asterisk-Users] Re: Realtime: I want to move the MySQL server out to anothermachine   Ronald Wiplinger
• [Asterisk-Users] I want to move the MySQL server out to another machine   Matthew Boehm
• [Asterisk-Users] Sip UA behind NAT   Wilson Pickett
• [Asterisk-Users] Polycom 500 'SERVICE'S' key   Chris Coulthurst
• [Asterisk-Users] Sounds   Giordano Grandis
• [Asterisk-Users] PRI Lines not being answered (No User Responding)   Tom Hayden
• [Asterisk-Users] No DTMF interpretation on outgoing calls   JunkMail
• [Asterisk-Users] I want to move the MySQL server out to another machine   Ronald Wiplinger
• [Asterisk-Users] Multiple E1s on one box   Jorge Alayon
• [Asterisk-Users] AMP and custom application   pellegrini at frameweb.it
• [Asterisk-Users] How to configure 2 asterisks   Jean-Michel Hiver
• [Asterisk-Users] te405p and dell poweredge   Daniel Nyström
• [Asterisk-Users] CallerID/chan_sccp   Julien Goodwin
• [Asterisk-Users] Queue Log   Hugo Begglo
• [Asterisk-Users] CallerID/chan_sccp   Sergio Chersovani
• [Asterisk-Users] AAH 1.1 - CRM Setup   Wiley Siler
• [Asterisk-Users] Re: SNOM extension lights programmable, eg. based on asterisk variable setting?   Josh Dady
• [Asterisk-Users] PRI Lines not being answered (No User Responding)   Francois Lambert
• [Asterisk-Users] realtime & nat   Victor Alvarez
• [Asterisk-Users] Monitor and failing Fax   John Knapp
• [Asterisk-Users] Re: *@home .conf files request   Iassen Hristov
• [Asterisk-Users] Issue with IAXy in Canada?   John Cianfarani
• [Asterisk-Users] realtime & nat   Zoa
• [Asterisk-Users] SS7   Matt
• [Asterisk-Users] SS7   Matt Fredrickson
• [Asterisk-Users] CallerID/chan_sccp   Joseph
• [Asterisk-Users] rxfax not answering   Antonio Gallo
• [Asterisk-Users] Re: SS7   Jason Stewart
• [Asterisk-Users] New Asterisk Manager Proxy -- astmanproxy 1.0   David C. Troy
• [Asterisk-Users] Re: SS7   Matt
• [Asterisk-Users] realtime & nat   Matthew Boehm
• [Asterisk-Users] Polycom 500 'SERVICE'S' key   Charlie Watts
• [Asterisk-Users] Call Routing based on number dialed (using S IP)   Geoff Manning
• [Asterisk-Users] rxfax not answering   pbx at itsngroup.com
• [Asterisk-Users] Polycom Phones & shorter than /24 netmasks   Charlie Watts
• [Asterisk-Users] CallerID/chan_sccp   Sergio Chersovani
• [Asterisk-Users] run a script on completion of call   C F
• [Asterisk-Users] Polycom Phones & shorter than /24 netmasks   Alexander Lopez
• [Asterisk-Users] Polycom 500 'SERVICE'S' key   Chris Coulthurst
• [Asterisk-Users] Polycom Phones & shorter than /24 netmasks   Chris Coulthurst
• [Asterisk-Users] Polycom Phones & shorter than /24 netmasks   Charlie Watts
• [Asterisk-Users] SS7   Matt
• [Asterisk-Users] AAH 1.1 - CRM Setup   David John Walsh
• [Asterisk-Users] Asterisk eating up 99.8% cpu   Moody
• [Asterisk-Users] AgentCallbackLogin (logout)   1 2
• [Asterisk-Users] Re: *@home .conf files request   Luis Diaz
• [Asterisk-Users] Servers Compatible with Digium HW   Jason Walker
• [Asterisk-Users] PRI Lines not being answered (No User Responding)   Tom Hayden
• [Asterisk-Users] RE: Asterisk-Users] te405p and dell poweredge   Jason Walker
• [Asterisk-Users] TE410P   Juan Pablo Abuyeres
• [Asterisk-Users] AAH 1.1 - CRM Setup   Wiley Siler
• [Asterisk-Users] TE410P   Tony Hoyle
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 48   Nguyen Trung Tin
• [Asterisk-Users] Problem in Reloading the asterisk server !   Zhu
• [Asterisk-Users] SPA-2002 and NAT   Waldo Rubinstein
• [Asterisk-Users] Queue Log   Johann
• [Asterisk-Users] TE410P   Matt Fredrickson
• [Asterisk-Users] TE410P   Juan Pablo Abuyeres
• [Asterisk-Users] TE410P   Brian McSpadden
• [Asterisk-Users] AgentCallbackLogin   Johann
• [Asterisk-Users] Polycom 500 'SERVICE'S' key   Johann
• [Asterisk-Users] run a script on completion of call   Johann
• [Asterisk-Users] TE410P   Juan Pablo Abuyeres
• [Asterisk-Users] X100P long delay before dial   Andres Paglayan
• [Asterisk-Users] TE410P   Juan Pablo Abuyeres
• [Asterisk-Users] Incorrect FAX detection.   Jon Califf
• [Asterisk-Users] Help! Zap echo on bridged calls   JD Austin
• [Asterisk-Users] Help! Zap echo on bridged calls   JD Austin
• [Asterisk-Users] DISA Help   Bill Madeira
• [Asterisk-Users] gxp-2000 tftp cfg   Peter Svensson
• [Asterisk-Users] DID on SIP channel   denis at isolve.com.br
• [Asterisk-Users] mISDN + chan_misdn.so + winbond issue   Michel Koenen
• [Asterisk-Users] Fax problem with Asterisk @home ver 1.0.7   Jose Vicente Ortega
• [Asterisk-Users] TE410P   Tony Hoyle
• [Asterisk-Users] connecting Asterisk to NEC NEAX system   Edwin Lam
• [Asterisk-Users] Answer Delay   Oswaldo Arratia
• [Asterisk-Users] Help! Zap echo on bridged calls   Kris Boutilier
• [Asterisk-Users] Fax problem with Asterisk @home ver 1.0.7   Wiley Siler
• [Asterisk-Users] AAH 1.1 - CRM Setup   Jeff Busch
• [Asterisk-Users] connecting Asterisk to NEC NEAX system   Dennis Walker
• [Asterisk-Users] DELL 2800 : PCI Parity error   David John Walsh
• [Asterisk-Users] ASTCC what has been changed   Ronald Wiplinger
• [Asterisk-Users] Gnudialer   Jesus Mogollon
• [Asterisk-Users] FXO Gateway recommendation   Adrian A
• [Asterisk-Users] Gnudialer   Chris A. Icide
• [Asterisk-Users] Interco H323 : IPNx (from WTL) and *   Eduardo Kaftanski
• [Asterisk-Users] gxp-2000 tftp cfg   David Phelan
• [Asterisk-Users] meetme recording of one user in the conference   Davin O'Neill
• [Asterisk-Users] Gnudialer   mattf
• [Asterisk-Users] ASTCC what has been changed   Darren Wiebe
• [Asterisk-Users] TE410P *SOLVED*   Juan Pablo Abuyeres
• [Asterisk-Users] codec preference   Mark Willis
• [Asterisk-Users] codec preference   Joshua Colp
• [Asterisk-Users] Answer Delay   Joshua Colp
• [Asterisk-Users] DISA Help   Joshua Colp
• [Asterisk-Users] Call Routing based on number dialed (using S IP)   Joshua Colp
• [Asterisk-Users] DID on SIP channel   Joshua Colp
• [Asterisk-Users] New Asterisk Manager Proxy -- astmanproxy 1.0   Joshua Colp
• [Asterisk-Users] Erro message - Received mini frame before firstfull voice frame   Joshua Colp
• [Asterisk-Users] Answer Delay   Oswaldo Arratia
• [Asterisk-Users] DID on SIP channel   Denis Galvão - iSolve
• [Asterisk-Users] DID on SIP channel   Joshua Colp
• [Asterisk-Users] 180 Ringing? (BUG?)   Joshua Colp
• [Asterisk-Users] Answer Delay   Joshua Colp
• [Asterisk-Users] Incoming voice "disappears"   Tony Davidson
• [Asterisk-Users] connecting Asterisk to NEC NEAX system   Edwin Lam
• [Asterisk-Users] DID on SIP channel   Denis Galvão - iSolve
• [Asterisk-Users] DID on SIP channel   Joshua Colp
• [Asterisk-Users] DID on SIP channel   Denis Galvão - iSolve
• [Asterisk-Users] Answer Delay   Oswaldo Arratia
• [Asterisk-Users] error message: INIT: Id "s0" respawning too fast: disable for 5 minutes   Don Wright
• [Asterisk-Users] Books   John H
• [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk   chawki hammoud
• [Asterisk-Users] DISA Help   Bill Madeira
• [Asterisk-Users] Books   Tony Davidson
• [Asterisk-Users] FXO Gateway recommendation   VoIP Newbie
• [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk   Ritesh Jalan
• [Asterisk-Users] so what are the additional hardware components needed?   infra struct
• [Asterisk-Users] Don't know what to do if second ROSE component is of type 0x6   Robert Goodyear
• [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk   chawki hammoud
• [Asterisk-Users] 180 Ringing? (BUG?)   Mirko Marghitola
• [Asterisk-Users] Message Playback   Sahil Gupta
• [Asterisk-Users] MGCP Useragent   Ibrar Ahmed
• [Asterisk-Users] rxfax not answering   Emanuele Pucciarelli
• [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk   Stefan Reuter
• [Asterisk-Users] FXO Gateway recommendation   Wai-Sun Chia
• [Asterisk-Users] MGCP Useragent   Florian Overkamp
• [Asterisk-Users] error message: INIT: Id "s0" respawning too fast:disable for 5 minutes   Florian Overkamp
• [Asterisk-Users] gxp-2000 tftp cfg   Peter Svensson
• [Asterisk-Users] Asterisk to Avaya PBX using TDM cards   Billy
• [Asterisk-Users] Newbie on asterisk ask for configuratio help   craz sead
• [Asterisk-Users] error message: INIT: Id "s0" respawning toofast:disable for 5 minutes   Wagner Gimenes
• [Asterisk-Users] bypass incoming ring..is it possible?   stevanus
• [Asterisk-Users] Xlite not communicating with Asterisk   Mohamed A. Gombolaty
• [Asterisk-Users] DISA Help   Wilson Pickett
• [Asterisk-Users] Xlite not communicating with Asterisk   Wilson Pickett
• [Asterisk-Users] bypass incoming ring..is it possible?   Alexander Ilyushin
• [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk   Alexander Ilyushin
• [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk   Ritesh Jalan
• [Asterisk-Users] Xlite not communicating with Asterisk   Shahan Kalutanthri
• [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk   Ritesh Jalan
• [Asterisk-Users] Message Playback   Alexander Ilyushin
• [Asterisk-Users] FXO Gateway recommendation   VoIP Newbie
• [Asterisk-Users] file.c:1073 ast_waitstream_full: Wait failed (Interrupted system call)   Mark Ackroyd
• [Asterisk-Users] Xlite not communicating with Asterisk   Zoa
• [Asterisk-Users] Xlite not communicating with Asterisk   Mohamed A. Gombolaty
• [Asterisk-Users] Xlite not communicating with Asterisk   Mohamed A. Gombolaty
• [Asterisk-Users] no DTMF pass-thru   Asterisk
• [Asterisk-Users] Xlite not communicating with Asterisk   Shahan Kalutanthri
• [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk   chawki hammoud
• [Asterisk-Users] performance of * in several scenarios   barney
• [Asterisk-Users] so what are the additional hardware componentsneeded?   Zoa
• [Asterisk-Users] Newbie on asterisk ask for configuratio help   f6hqz-m at hamwlan.net
• [Asterisk-Users] SPA-2002 and NAT   Chris Mason (Lists)
• [Asterisk-Users] DID on SIP channel   Olle E. Johansson
• [Asterisk-Users] How to handle one incoming call on multiple lines?   Erwin Lubbers
• [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received   Marco Parmeggiani
• [Asterisk-Users] Faxing error rtp.c:504 ast_rtp_read: Unknown RTP codec 100 received   Chris Mason (Lists)
• [Asterisk-Users] English vs American voice files   Andrew Thrift
• [Asterisk-Users] DID on SIP channel   Olle E. Johansson
• [Asterisk-Users] AT-320 + supervised transfer   Gavin Hamill
• [Asterisk-Users] so what are the additional hardware componentsneeded?   Steve Totaro
• [Asterisk-Users] Station Lines   Sean Cook
• [Asterisk-Users] D-link DPH-80 (SIP) call to asterisk problem   Eugene Crosser
• [Asterisk-Users] newbie question   Charles Austin
• [Asterisk-Users] How to handle one incoming call on multiple lines?   Julian J. M.
• [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk   Marcelo Pacheco
• [Asterisk-Users] IAXtel update!   Kevin P. Fleming
• [Asterisk-Users] Books   The VoIP Connection
• [Asterisk-Users] Station Lines   Andrew Kohlsmith
• [Asterisk-Users] English vs American voice files   Paul Redstone
• [Asterisk-Users] SS7   Kevin P. Fleming
• [Asterisk-Users] English vs American voice files   Mark Phillips
• [Asterisk-Users] Asterisk to Avaya PBX using TDM cards   Rich Adamson
• [Asterisk-Users] English vs American voice files   Sahil Gupta
• [Asterisk-Users] error message: INIT: Id "s0" respawning toofast:disable for 5 minutes   Justin Ellison
• [Asterisk-Users] CallerID/chan_sccp   Sergio Chersovani
• [Asterisk-Users] Station Lines   Walt Reed
• [Asterisk-Users] Fax + Fritz + Capi + detection   sylvain garcia
• [Asterisk-Users] Books   Zoa
• [Asterisk-Users] Polycom 500 "Group Call Pickup Feature" and *   Chris Coulthurst
• [Asterisk-Users] sip to sip echo with meetme, timing   Jerry Bonner
• [Asterisk-Users] * @ Home: All Circuits busy   maoleson at mchsi.com
• [Asterisk-Users] How to handle one incoming call on multiple lines?   Erwin Lubbers
• [Asterisk-Users] How to handle one incoming call on multiple lines?   Andrew Latham
• [Asterisk-Users] * @ Home: All Circuits busy   Dan Littlejohn
• [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch   Neil and Fiona
• [Asterisk-Users] How to handle one incoming call on multiple lines?   Florian Overkamp
• [Asterisk-Users] Latest CVS and app_rxfax   Dave Cotton
• [Asterisk-Users] * @ Home: All Circuits busy   Dean Mumby
• [Asterisk-Users] Asterisk and Alcatel 4200 PBX   José Luis Gómez
• [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk   chawki hammoud
• [Asterisk-Users] newbie question   Moises Silva
• [Asterisk-Users] Clicks in audio with TE100P PRI   Alejandro G
• [Asterisk-Users] no DTMF pass-thru   Moises Silva
• [Asterisk-Users] How to handle one incoming call on multiple lines?   Henry Coleman
• [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch   Andrew Kohlsmith
• [Asterisk-Users] rxfax not answering   JD Austin
• [Asterisk-Users] * @ Home: All Circuits busy   maoleson at mchsi.com
• [Asterisk-Users] Clicks in audio with TE100P PRI   Andrew Kohlsmith
• [Asterisk-Users] How to handle one incoming call on multiple lines?   Henry Coleman
• [Asterisk-Users] How to handle one incoming call on multiple lines?   Andrew Kohlsmith
• [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch   Neil and Fiona
• [Asterisk-Users] CVS Head, Flex 2.5.31 or higher? READ THIS!   Steve Murphy
• [Asterisk-Users] CallerID/chan_sccp   Julien Goodwin
• [Asterisk-Users] Remote CDR logging on mysql:   Tim Connolly
• [Asterisk-Users] How to handle one incoming call on multiple lines?   Henry Coleman
• [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch   Andrew Kohlsmith
• [Asterisk-Users] Remote CDR logging on mysql:   Matthew Boehm
• [Asterisk-Users] Queue Log   Hugo Begglo
• [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch   Julian J. M.
• [Asterisk-Users] TDM04B   anderson at skyweb.co.ke
• [Asterisk-Users] CallerID/chan_sccp   Joseph
• [Asterisk-Users] Echo problem   Martin Roy
• [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk   Leandro Tenorio
• [Asterisk-Users] Help Connecting Cisco AS5300 to Asterisk   Leandro Tenorio
• [Asterisk-Users] EuroISDN Italy - quadbri - zaptel.conf - what settings work ?   Robert Rozman
• [Asterisk-Users] Echo problem   Michael D Schelin
• [Asterisk-Users] Number of AGI's running at the same time   Jerry Geis
• [Asterisk-Users] rxfax not working   Jay Austad
• [Asterisk-Users] Echo problem   Andrew Kohlsmith
• [Asterisk-Users] Queue Log   Brian Roy
• [Asterisk-Users] * @ Home: All Circuits busy   Asterisk at Home
• [Asterisk-Users] Echo problem   Jon Califf
• [Asterisk-Users] Multiple E1s on one box   Franco Bellagamba
• [Asterisk-Users] * @ Home: All Circuits busy   Greg Jones Media
• [Asterisk-Users] * @ Home: All Circuits busy   maoleson at mchsi.com
• [Asterisk-Users] Do I need a ring capacitor to use TDM400P cards in UK   Angus Comber
• [Asterisk-Users] English vs American voice files   Mike M
• [Asterisk-Users] * @ Home: All Circuits busy   asterisk at txpe.net
• [Asterisk-Users] * @ Home: All Circuits busy   Dan Littlejohn
• [Asterisk-Users] Echo problem   Michael D Schelin
• [Asterisk-Users] hardware question   Michel Brabants
• [Asterisk-Users] * @ Home: All Circuits busy   Dean Mumby
• [Asterisk-Users] How to handle one incoming call on multiple lines?   Greg Oliver
• [Asterisk-Users] IP PHONE iareaphone x100, tested??   Jorge Ortega Perez
• [Asterisk-Users] rxfax not working   Jay Austad
• [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch   Rich Adamson
• [Asterisk-Users] Multiple E1s on one box   Moises Silva
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Roman Zhovtulya
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Scott Wolfe
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Julio Arruda
• [Asterisk-Users] IP PHONE iareaphone x100, tested??   snacktime
• [Asterisk-Users] IP PHONE iareaphone x100, tested??   Mark Brown
• [Asterisk-Users] * @ Home: All Circuits busy   maoleson at mchsi.com
• [Asterisk-Users] Load per server?   Bryan Field-Elliot
• [Asterisk-Users] ENUM NL dead ?   Michiel van Baak
• [Asterisk-Users] Echo problem   Rich Adamson
• [Asterisk-Users] Multiple E1s on one box   Jorge Alayon
• [Asterisk-Users] Do I need a ring capacitor to use TDM400P cards in UK   Henry Coleman
• [Asterisk-Users] * @ Home: All Circuits busy   Dean Mumby
• [Asterisk-Users] IP PHONE iareaphone x100, tested??   Kanuri, Seshu (Company IT)
• [Asterisk-Users] IP PHONE iareaphone x100, tested??   Kanuri, Seshu (Company IT)
• [Asterisk-Users] AgentCallBacklogin (logout continued...)   1 2
• [Asterisk-Users] Multiple E1s on one box   Rusty Shackleford
• [Asterisk-Users] Do I need a ring capacitor to use TDM400P cards in UK   David John Walsh
• [Asterisk-Users] format g729 and Voxee.com   Todd Reese
• [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch   Neil and Fiona
• [Asterisk-Users] Little help with MySQL please   Time Bandit
• [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch   Neil and Fiona
• [Asterisk-Users] Incoming call stops at random with Teliax   Rick Baranowski
• [Asterisk-Users] Echo problem   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Cisco 7960 mic generating noise on other end   Lex Lethol
• [Asterisk-Users] Erro message - Received mini frame before firstfull voice frame   Adam Goryachev
• [Asterisk-Users] Echo problem   Kris Boutilier
• [Asterisk-Users] Ringing a few phones   Shidan
• [Asterisk-Users] Echo problem   Neil and Fiona
• [Asterisk-Users] Ringing a few phones   Jennifer Hales
• [Asterisk-Users] Group/Broadcast Voicemail   Chris Stinson
• [Asterisk-Users] More than one account from the same provider?   Chris Mason (Lists)
• [Asterisk-Users] bypass incoming ring..is it possible?   stevanus
• [Asterisk-Users] Ringing a few phones   Shidan
• [Asterisk-Users] Group/Broadcast Voicemail   Robert Goodyear
• [Asterisk-Users] More than one account from the same provider?   Robert Goodyear
• [Asterisk-Users] ivr not working?   stevanus
• [Asterisk-Users] Ringing a few phones   Robert Goodyear
• [Asterisk-Users] [ADMIN]: subscription failure   David Koski
• [Asterisk-Users] Group/Broadcast Voicemail   Chris Stinson
• [Asterisk-Users] Clicks in audio with TE100P PRI   Alejandro G
• [Asterisk-Users] Clicks in audio with TE100P PRI   Adam Goryachev
• [Asterisk-Users] Group/Broadcast Voicemail   Robert Goodyear
• [Asterisk-Users] Clicks in audio with TE100P PRI   James Bean
• [Asterisk-Users] tdm04b slow response   stevanus
• [Asterisk-Users] GXP2000 and hint LED's   James Bean
• [Asterisk-Users] tdm04b slow response   Neil and Fiona
• [Asterisk-Users] Play MP3 during Record   Phuong Nguyen
• [Asterisk-Users] GXP2000 and hint LED's   Peter Svensson
• [Asterisk-Users] hardware question   Wilson Pickett
• [Asterisk-Users] Thank you for the timely suggestion   infra struct
• [Asterisk-Users] IP PHONE iareaphone x100, tested??   Wilson Pickett
• [Asterisk-Users] bypass incoming ring..is it possible?   Wilson Pickett
• [Asterisk-Users] Thank you for the timely suggestion   stevanus
• [Asterisk-Users] Asterisk Engineer/Programmer required   Motavi
• [Asterisk-Users] TDM400P strangeness   Jean-Michel Hiver
• [Asterisk-Users] bypass incoming ring..is it possible?   stevanus
• [Asterisk-Users] hardware question   Henry Jensen
• [Asterisk-Users] GXP2000 and hint LED's   James Bean
• [Asterisk-Users] performance of * in several scenarios   barney
• [Asterisk-Users] Notice Message   craz sead
• [Asterisk-Users] Re: mISDN + chan_misdn.so + winbond issue   Michel Koenen
• [Asterisk-Users] TDM400P strangeness   Jean-Denis Girard
• [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch   Julian J. M.
• [Asterisk-Users] New version 1.013 of Asterisk VConfig   snacktime
• [Asterisk-Users] EuroISDN Italy - quadbri - zaptel.conf - what settings work ?   Matteo Brancaleoni
• [Asterisk-Users] Softphone for Linux desktops   Eric Bishop
• [Asterisk-Users] Softphone for Linux desktops   stevanus
• [Asterisk-Users] Softphone for Linux desktops   Florian Overkamp
• [Asterisk-Users] Do I need a ring capacitor to use TDM400P cards in UK   Mark Elkins
• [Asterisk-Users] Clicks in audio with TE100P PRI   Andrew Kohlsmith
• [Asterisk-Users] TDM04B   anderson at skyweb.co.ke
• [Asterisk-Users] TDM04B   Andrew Kohlsmith
• [Asterisk-Users] Asterisk to Cisco Unity   Simone
• [Asterisk-Users] 180 Ringing? (BUG?)   Mirko Marghitola
• [Asterisk-Users] More than one account from the same provider?   Chris Mason (Lists)
• [Asterisk-Users] TDM04B   Wilson Pickett
• [Asterisk-Users] Pickup problem   Kib Eki
• [Asterisk-Users] Clicks in audio with TE100P PRI   Peter Svensson
• [Asterisk-Users] TDM04B   Ariel Batista
• [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch   Neil and Fiona
• [Asterisk-Users] OT: Please comment on Dvorak's troll   Charles Austin
• [Asterisk-Users] 3COM NBX SuperStack 3   Martin Croome
• [Asterisk-Users] TDM04B   anderson at skyweb.co.ke
• [Asterisk-Users] More than one account from the same provider?   Chris Mason (Lists)
• [Asterisk-Users] Incoming call stops at random with Teliax   Rich Adamson
• [Asterisk-Users] * @ Home: All Circuits busy   Greg Jones Media
• [Asterisk-Users] More than one account from the same provider?   Rich Adamson
• [Asterisk-Users] [ADMIN]: subscription failure   Walt Reed
• [Asterisk-Users] format g729 and Voxee.com   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Play MP3 during Record   Brian Roy
• [Asterisk-Users] TDM04B   Rich Adamson
• [Asterisk-Users] More than one account from the same provider?   Chris Mason (Lists)
• [Asterisk-Users] format g729 and Voxee.com   Mattt
• [Asterisk-Users] Softphone for Linux desktops   Jason Becker
• [Asterisk-Users] Lingo(.com) and Asterisk   Bas Rijniersce
• [Asterisk-Users] Re: format g729 and Voxee.com   Caleb
• [Asterisk-Users] format g729 and Voxee.com   Rich Adamson
• [Asterisk-Users] 3COM NBX SuperStack 3   Chris Hills
• [Asterisk-Users] More than one account from the same provider?   Rich Adamson
• [Asterisk-Users] 180 Ringing? (BUG?)   Julian J. M.
• [Asterisk-Users] GXP2000 and hint LED's   Julian J. M.
• [Asterisk-Users] Group/Broadcast Voicemail   Chris Stinson
• [Asterisk-Users] Lingo(.com) and Asterisk   Colin Anderson
• [Asterisk-Users] TDM400P... ignoring hanguponpolarityswitch   Neil and Fiona
• [Asterisk-Users] Incoming call stops at random with Teliax   Rick Baranowski
• [Asterisk-Users] having to reload asterix after internet connection failure   Armand Sulter
• [Asterisk-Users] E1 and SS7   Michael Welter
• [Asterisk-Users] DID on SIP channel   Denis Galvão - iSolve
• [Asterisk-Users] Cisco 7960 and Skinny   Stojan Sljivic - GDS
• [Asterisk-Users] Cisco 7960 and Skinny   Sergio Chersovani
• [Asterisk-Users] Re: AgentCallBacklogin (logout continued...)   alan
• [Asterisk-Users] REPOSTED: Polycom 500 "Group Call Pickup Feature" and *   Chris Coulthurst
• [Asterisk-Users] howto write CDRs on two mysql servers   Rosario Pingaro
• [Asterisk-Users] DISA Help   Bill Madeira
• [Asterisk-Users] CallerID/chan_sccp   Mark Johnson
• [Asterisk-Users] Polycom IP-500 & 600 Nat settings.   Ariel Batista
• [Asterisk-Users] Clicks in audio with TE100P PRI   Matt Fredrickson
• [Asterisk-Users] Group/Broadcast Voicemail   Robert Goodyear
• [Asterisk-Users] astGUIclient installation problem   kritikus Araklidas
• [Asterisk-Users] howto write CDRs on two mysql servers   Matthew Boehm
• [Asterisk-Users] More than one account from the same provider?   Robert Goodyear
• [Asterisk-Users] CallerID/chan_sccp   Sergio Chersovani
• [Asterisk-Users] Asterisk to Cisco Unity   Simone
• [Asterisk-Users] having to reload asterix after internet connection failure   Moises Silva
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Pedro
• [Asterisk-Users] E1 and SS7   VOIP Consultant
• [Asterisk-Users] Asterisk Live! CF   abel
• [Asterisk-Users] More than one account from the same provider?   Rich Adamson
• [Asterisk-Users] having to reload asterix after internet connection failure   Dave Cotton
• [Asterisk-Users] Notice Message   Johann
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Wiley Siler
• [Asterisk-Users] Agent refuses to log out   Asterisk
• [Asterisk-Users] having to reload asterix after internet connection failure   Rich Adamson
• [Asterisk-Users] Asterisk Live! CF   Kristian Kielhofner
• [Asterisk-Users] Incoming call stops at random with Teliax   Rich Adamson
• [Asterisk-Users] Asterisk Live! CF   Bob Goddard
• [Asterisk-Users] having to reload asterix after internet connection failure   Armand Sulter
• [Asterisk-Users] howto write CDRs on two mysql servers   Robert Goodyear
• [Asterisk-Users] astGUIclient installation problem   mattf
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   George Pajari
• [Asterisk-Users] howto write CDRs on two mysql servers   Mark Musone
• [Asterisk-Users] RE: astGUIclient installation problem   David Gomillion
• [Asterisk-Users] Asterisk to Cisco Unity   William Boehlke
• [Asterisk-Users] Lingo(.com) and Asterisk   Luki
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Kris Boutilier
• [Asterisk-Users] Asterisk to Cisco Voip System Unity   Simone
• [Asterisk-Users] ATTN: Keith   Matt
• [Asterisk-Users] howto write CDRs on two mysql servers   Andrew Latham
• [Asterisk-Users] Digium vs. Sangoma: Performance   mattf
• [Asterisk-Users] Clicks in audio with TE100P PRI   Zoa
• [Asterisk-Users] having to reload asterix after internet connection failure   Wilson Pickett
• [Asterisk-Users] Asterisk Live! CF   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Re: Lingo(.com) and Asterisk   M O
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Wiley Siler
• [Asterisk-Users] REPOSTED: Polycom 500 "Group Call Pickup Feature" and *   Olle E. Johansson
• [Asterisk-Users] E1 and SS7   Mike M
• [Asterisk-Users] OT: SpamFiltering (used to be: ATTN: Keith)   Wiley Siler
• [Asterisk-Users] CallerID/chan_sccp   Joseph
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Rusty Shackleford
• [Asterisk-Users] howto write CDRs on two mysql servers   Rusty Shackleford
• [Asterisk-Users] ATTN: Keith   Sean Kennedy
• [Asterisk-Users] Getting a Quintum AS200 to connect with Asterisk using SIP?   Min Hwan Chang
• [Asterisk-Users] howto write CDRs on two mysql servers   Matthew Boehm
• [Asterisk-Users] howto write CDRs on two mysql servers   Huddleston, Robert
• [Asterisk-Users] Asterisk Live! CF   Bob Goddard
• [Asterisk-Users] CallerID/chan_sccp   Mark Johnson
• [Asterisk-Users] GXP2000 and hint LED's   Peter Svensson
• [Asterisk-Users] Queue Log   Hugo Begglo
• [Asterisk-Users] Asterisk to Cisco Unity   Rich Adamson
• [Asterisk-Users] Asterisk to Cisco Unity   info at beprojects.com
• [Asterisk-Users] Lingo(.com) and Asterisk   Dean Collins
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Race Vanderdecken
• [Asterisk-Users] ATTN: Keith   Dean Collins
• [Asterisk-Users] Group/Broadcast Voicemail   Chris Stinson
• [Asterisk-Users] Inbound provider in Canada   Adrian A
• [Asterisk-Users] [ADMIN]: subscription failure   David Koski
• [Asterisk-Users] ATTN: Keith   list
• [Asterisk-Users] GXP2000 and hint LED's   Michiel van Baak
• [Asterisk-Users] Voicemail and MS Exchange   magnus
• [Asterisk-Users] Asterisk Live! CF   The VoIP Connection
• [Asterisk-Users] voicemail check for busy message   Jim Sturtevant
• [Asterisk-Users] Phantom (ghost) Calls with Wildcard TDM400P   John Lange
• [Asterisk-Users] ATTN: Keith   Huddleston, Robert
• [Asterisk-Users] Play MP3 during Record   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] GXP2000 and hint LED's   Peter Svensson
• [Asterisk-Users] GXP2000 and hint LED's   James Bean
• [Asterisk-Users] GXP2000 and hint LED's   Michiel van Baak
• [Asterisk-Users] Parked Call queue function key notify   James Bean
• [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization   Iassen Hristov
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   support at sjobeck.com
• [Asterisk-Users] voicemail check for busy message   Chris Coulthurst
• [Asterisk-Users] VOIP-INFO   Chris Coulthurst
• [Asterisk-Users] Voicemail and MS Exchange   support at sjobeck.com
• [Asterisk-Users] VOIP-INFO   Dan Perik
• [Asterisk-Users] VOIP-INFO   Bryce Chidester
• [Asterisk-Users] VOIP-INFO   Wiley Siler
• [Asterisk-Users] Group/Broadcast Voicemail   Robert Goodyear
• [Asterisk-Users] GXP2000 and hint LED's   The VoIP Connection
• [Asterisk-Users] VOIP-INFO   Neal Walton
• [Asterisk-Users] VOIP-INFO   Chris Coulthurst
• [Asterisk-Users] VOIP-INFO.ORG   Michiel van Baak
• [Asterisk-Users] Group/Broadcast Voicemail   Robert Goodyear
• [Asterisk-Users] TDM04B   Andrew Kohlsmith
• [Asterisk-Users] TDM04B   Andrew Kohlsmith
• [Asterisk-Users] Clicks in audio with TE100P PRI   Andrew Kohlsmith
• [Asterisk-Users] Voicemail and MS Exchange   C F
• [Asterisk-Users] Cisco 7960 and Skinny   asterisk at marnock.net
• [Asterisk-Users] Flash Hook won't work with Asterisk@Home and SPS-2002   Todd A. Riker
• [Asterisk-Users] Asterisk Live! CF   Andrew Kohlsmith
• [Asterisk-Users] Inbound provider in Canada   MF Hulber
• [Asterisk-Users] Re: Lingo(.com) and Asterisk   M O
• [Asterisk-Users] Inbound provider in Canada   Darren Wiebe
• [Asterisk-Users] GXP2000 and hint LED's   The VoIP Connection
• [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization   C F
• [Asterisk-Users] ASTCC what has been changed   Ronald Wiplinger
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   C F
• [Asterisk-Users] VOIP-INFO   Ryan Finnesey
• [Asterisk-Users] E1 and SS7   Michael Welter
• [Asterisk-Users] Group/Broadcast Voicemail   Chris Stinson
• [Asterisk-Users] Sixtel is still alive?   Jay Milk
• [Asterisk-Users] Issue with IAXy in Canada?   Obaid Siddiqui
• [Asterisk-Users] Handytone-488 FXO - PSTN in calls to Asterisk, sip.conf?   Albert Lash
• [Asterisk-Users] GXP2000 and hint LED's   James Bean
• [Asterisk-Users] GXP-2000 Wiki update..   Rob Thomas
• [Asterisk-Users] Sixtel is still alive?   Hadar Pedhazur
• [Asterisk-Users] VOIP-INFO   Albert Lash
• [Asterisk-Users] E1 and SS7   Mike M
• [Asterisk-Users] Multiple Digium cards?   geek at mnwifi.net
• [Asterisk-Users] Phantom (ghost) Calls with Wildcard TDM400P   Steve Totaro
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Craig Guy
• [Asterisk-Users] VOIP-INFO   Peter A. Solomon
• [Asterisk-Users] compile error cannot find -lidn   Kamran Ahmad
• [Asterisk-Users] compile error cannot find -lidn   Tom Lanyon
• [Asterisk-Users] Conversations cuts: "didn't get a frame from Channel: SIP/..."   Juan Manuel Coronado Z.
• [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization   James Treleaven
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Moody
• [Asterisk-Users] "auto-dial out" not waiting for answer   James Treleaven
• [Asterisk-Users] [Asterisk-biz] Asterisk Evening in Melbourne (again!) next Thursday   jurgen
• [Asterisk-Users] ASTCC what has been changed   Darren Wiebe
• [Asterisk-Users] Voicemail and MS Exchange   Race Vanderdecken
• [Asterisk-Users] VOIP-INFO   Steve Totaro
• [Asterisk-Users] Voicemail and MS Exchange   Ryan Finnesey
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Tim Litwiller
• [Asterisk-Users] VOIP-INFO   Peter A. Solomon
• [Asterisk-Users] VOIP-INFO   Chris Coulthurst
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Kris Boutilier
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Colin Anderson
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Tim Litwiller
• [Asterisk-Users] VOIP-INFO   Peter A. Solomon
• [Asterisk-Users] IAX2 Max Retries dropped calls Firefly   Paul Redstone
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Chris Coulthurst
• [Asterisk-Users] GXP2000 and hint LED's   Peter Svensson
• [Asterisk-Users] GXP2000 and hint LED's   Peter Svensson
• [Asterisk-Users] GXP2000 and hint LED's   Peter Svensson
• [Asterisk-Users] VOIP-INFO   James H. Thompson
• [Asterisk-Users] GXP2000 and hint LED's   Peter Svensson
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   support at sjobeck.com
• [Asterisk-Users] Asterisk Live! CF   Bob Goddard
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Dan Levine
• [Asterisk-Users] Is it possible to have a remote Phone work behind Nat without a VPN?   Dan Levine
• [Asterisk-Users] VOIP-INFO   Olle E. Johansson
• [Asterisk-Users] Zap Clocking - Frame Slips - tdm400p wcfxo zttest cpu spikes spandsp   qrss
• [Asterisk-Users] IAX2 Max Retries dropped calls Firefly   Adam Hart
• [Asterisk-Users] PHPAGI Swift Escape Digits   Michael Stearne
• [Asterisk-Users] Play MP3 during Record   Phuong Nguyen
• [Asterisk-Users] ENUM NL dead ?   Florian Overkamp
• [Asterisk-Users] sirrix NT mode   altus
• [Asterisk-Users] GXP2000 and hint LED's   The VoIP Connection
• [Asterisk-Users] GXP2000 and hint LED's   Peter Svensson
• [Asterisk-Users] TDM04B   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] ATTN: Keith   Dave Cotton
• [Asterisk-Users] ATTN: Keith - Seriously OT   Terry H. Gilsenan
• [Asterisk-Users] G.729AB codec support   Erdem HAKİ
• [Asterisk-Users] Request OPTION and 404 Sjphone Xlite   sylvain garcia
• [Asterisk-Users] lost g729 lic   altus
• [Asterisk-Users] Asterisk to Cisco Unity   Simone
• [Asterisk-Users] G.729AB codec support   Soner Tari
• [Asterisk-Users] help for conference   craz sead
• [Asterisk-Users] G.729AB codec support   Zoa
• [Asterisk-Users] G.729AB codec support   Soner Tari
• [Asterisk-Users] G.729AB codec support   Zoa
• [Asterisk-Users] lost g729 lic   Andrew Kohlsmith
• [Asterisk-Users] VOIP-INFO   Andrew Kohlsmith
• [Asterisk-Users] ATTN: Keith - Seriously OT   Andrew Kohlsmith
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Andrew Kohlsmith
• [Asterisk-Users] Call inband progress indication and zaphfc   Diego Ercolani
• [Asterisk-Users] Asterisk Live! CF   Andrew Kohlsmith
• [Asterisk-Users] lost g729 lic   altus
• [Asterisk-Users] Request OPTION and 404 Sjphone Xlite   Olle E. Johansson
• [Asterisk-Users] SIP Authentication   Stojan Sljivic - GDS
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Guido Hecken
• [Asterisk-Users] ATTN: Keith - Seriously OT   Terry H. Gilsenan
• [Asterisk-Users] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU   Andres Maduro
• [Asterisk-Users] Handytone-488 FXO - PSTN in calls to Asterisk, sip.conf? (fwd)   Albert Lash
• [Asterisk-Users] ATTN: Keith - Seriously OT   Andrew Kohlsmith
• [Asterisk-Users] Cell redirect   hugoboy at inbox.lv
• [Asterisk-Users] TE410P and Siemens HIPATH 3750   Sergio Serrano
• [Asterisk-Users] VOIP-INFO   Chris Coulthurst
• [Asterisk-Users] SpanDSP wownt compile   Mark Ratering
• [Asterisk-Users] VOIP-INFO   trixter http://www.0xdecafbad.com
• [Asterisk-Users] g729 support   Erdem HAKİ
• [Asterisk-Users] Asterisk Evening in Melbourne (again!) next Thursday   Huddleston, Robert
• [Asterisk-Users] VOIP-INFO   Nir Simionovich
• [Asterisk-Users] Is it possible to have a remote Phone work behind Nat without a VPN?   Durf
• [Asterisk-Users] re: PHPAGI Swift Escape Digits   Clarke Kawakami
• [Asterisk-Users] VOIP-INFO   Nathan Pralle
• [Asterisk-Users] Asterisk to Cisco Unity   Peter Braidwood
• [Asterisk-Users] Channel Banks   David Sampson
• [Asterisk-Users] Zap Clocking - Frame Slips - tdm400p wcfxo zttest cpu spikes spandsp   Rich Adamson
• [Asterisk-Users] 404 not found   sylvain garcia
• [Asterisk-Users] SoftPhone - Solaris   Sebastian Silva
• [Asterisk-Users] ATTN: Keith - Seriously OT   Neal Walton
• [Asterisk-Users] TDM04B   David Brodbeck
• [Asterisk-Users] Voicemail and MS Exchange   David Brodbeck
• [Asterisk-Users] VOIP-INFO   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Re: Voicemail and MS Exchange Synchronizatio n   David Brodbeck
• [Asterisk-Users] help for conference   Moises Silva
• [Asterisk-Users] TE410P and Siemens HIPATH 3750   Henry Jensen
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Pedro
• [Asterisk-Users] D-Link DVG-1402S   Luis Czop
• [Asterisk-Users] Is it possible to have a remote Phone work behind Nat without a VPN?   Dan Levine
• [Asterisk-Users] Is it possible to have a remote Phone work behindNat without a VPN?   Maxime Renaud
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 69   Nguyen Trung Tin
• [Asterisk-Users] asterisk and mpg123 lock up   Jerry Geis
• [Asterisk-Users] Channel Banks   Andrew Latham
• [Asterisk-Users] config problem   Georges Henroteaux
• [Asterisk-Users] Clicks in audio with TE100P PRI   Alejandro G
• [Asterisk-Users] config problem   Moises Silva
• [Asterisk-Users] Best BootRom & SIP Code for Poly600?   Justin Ellison
• [Asterisk-Users] AAH 1.1 cannot call between extensions (xten lite softphones)   Nick Heinemans
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Matt
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Matt
• [Asterisk-Users] G711 ( alaw or ulaw ) pass-thru   Edgardo Bermejo
• [Asterisk-Users] Clicks in audio with TE100P PRI   Kris Boutilier
• [Asterisk-Users] G711 ( alaw or ulaw ) pass-thru   Sahil Gupta
• [Asterisk-Users] Re: Best BootRom & SIP Code for Poly600?   Noah Miller
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Pedro
• [Asterisk-Users] ASTCC what has been changed   Ronald Wiplinger
• [Asterisk-Users] blindtransfers with IAX   Marc Storck
• [Asterisk-Users] Dropping Frame of G729   Matthew Boehm
• [Asterisk-Users] G711 ( alaw or ulaw ) pass-thru   Steve Underwood
• [Asterisk-Users] Cell redirect   Ugis Racko
• [Asterisk-Users] re: PHPAGI Swift Escape Digits   Michael Stearne
• [Asterisk-Users] Clicks in audio with TE100P PRI   Alejandro G
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Race Vanderdecken
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 71   Nguyen Trung Tin
• [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization   Race Vanderdecken
• [Asterisk-Users] Clicks in audio with TE100P PRI   Kris Boutilier
• [Asterisk-Users] Call disconnect   Scott England
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   support at sjobeck.com
• [Asterisk-Users] Cell redirect   Moises Silva
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Roman Zhovtulya
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Roman Zhovtulya
• [Asterisk-Users] Best BootRom & SIP Code for Poly600?   Ariel Batista
• [Asterisk-Users] Best BootRom & SIP Code for Poly600?   Tarpo, Louie
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Wiley Siler
• [Asterisk-Users] Toll Free DIDs   Hugh L. Johnson
• [Asterisk-Users] Toll Free DIDs   John Millican
• [Asterisk-Users] Toll Free DIDs   Paul Traue, Jr.
• [Asterisk-Users] Phantom (ghost) Calls with Wildcard TDM400P   John Lange
• [Asterisk-Users] Phantom (ghost) Calls with Wildcard TDM400P   jltaylor
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Chris Mason (Lists)
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Wiley Siler
• [Asterisk-Users] ATTN: Keith   Sean Kennedy
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   David Brodbeck
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Dean Collins
• [Asterisk-Users] Unable To Register A SIP phone   SYED ADEEL ALI
• [Asterisk-Users] ATTN: Keith   Mark Musone
• [Asterisk-Users] DMS-500 CID name not in CDR   jr.richardson at cox.net
• [Asterisk-Users] Unable to register Zyxel WIFI Phone as SIP Client to Asterisk   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Newie Questions   Matthew T. O'Connor
• [Asterisk-Users] Newie Questions   Dean Collins
• [Asterisk-Users] ATTN: Keith   list
• [Asterisk-Users] Convert extensions.conf INTO MySQL script   Chris Coulthurst
• [Asterisk-Users] DMS-500 CID name not in CDR   David Boyd
• [Asterisk-Users] Polycoms Go Silent after a a handful of calls.   Ryan Stark
• [Asterisk-Users] voiceblue gsm/sip   Trey Scarborough
• [Asterisk-Users] Polycoms Go Silent after a a handful of calls.   Joshua Stephens
• [Asterisk-Users] VOIP-INFO   Andrew Kohlsmith
• [Asterisk-Users] Cisco 7960 mic generating noise on other end   cano gackt
• [Asterisk-Users] Convert extensions.conf INTO MySQL script   Matthew Boehm
• [Asterisk-Users] VOIP-INFO   Peter A. Solomon
• [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization   Iassen Hristov
• [Asterisk-Users] Open telecoms.org   Roger Schreiter
• [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization   Steve Kennedy
• [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization   Iassen Hristov
• [Asterisk-Users] Sixtel is still alive?   Simon Coles
• [Asterisk-Users] Wildly inaccurate CDR records   Obelix
• [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization   Race Vanderdecken
• [Asterisk-Users] DHCP vendor strings   Kevin Kiely
• [Asterisk-Users] Re:DMS-500 CID name not in CDR   JR Richardson
• [Asterisk-Users] ASTCC what has been changed   Darren Wiebe
• [Asterisk-Users] First Asterisk community meeting in Sydney   Shamsul Arefin
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Barton Fisher
• [Asterisk-Users] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU   Steve Underwood
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Michael Welter
• [Asterisk-Users] G.729AB codec support   stevanus
• [Asterisk-Users] [newbie] configuration for IAX server to server   David Koski
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Tom
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Robert Goodyear
• [Asterisk-Users] G.729AB codec support   Steve Underwood
• [Asterisk-Users] Asterisk@Home connecting through firewall/router   C. Hatton Humphrey
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Rick Baranowski
• [Asterisk-Users] Asterisk@Home connecting through firewall/router   Peter A. Solomon
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Peter A. Solomon
• [Asterisk-Users] [newbie] configuration for IAX server to server   Shamsul Arefin
• [Asterisk-Users] Newie Questions   Matthew T. O'Connor
• [Asterisk-Users] VOIP-INFO   Andrew Kohlsmith
• [Asterisk-Users] Is it necessary that i need to have TDM01B for PC-to-PC intercom calls?   infra struct
• [Asterisk-Users] Is it necessary that i need to have TDM01B for PC-to-PC intercom calls?   infra struct
• [Asterisk-Users] VOIP-INFO   trixter http://www.0xdecafbad.com
• [Asterisk-Users] [newbie] configuration for IAX server to server   David Koski
• [Asterisk-Users] VoicePulse DTMF Problems Anyone?   Michael Stearne
• [Asterisk-Users] what is asteriskathome-1.0.iso?   infra struct
• [Asterisk-Users] what is asteriskathome-1.0.iso?   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Newie Questions   Rich Adamson
• [Asterisk-Users] VoicePulse DTMF Problems Anyone?   Michael Stearne
• [Asterisk-Users] VoicePulse DTMF Problems Anyone?   Rich Adamson
• [Asterisk-Users] Newbie Here..... Unable To Register A SIP phone   SYED ADEEL ALI
• [Asterisk-Users] Newie Questions RE: Polycom Critique   Chris Coulthurst
• [Asterisk-Users] How to configure Asterisk as sip proxy   Ibrar Ahmed
• [Asterisk-Users] VoicePulse DTMF Problems Anyone?   Michael Stearne
• [Asterisk-Users] VoicePulse DTMF Problems Anyone?   Rich Adamson
• [Asterisk-Users] ATTN: Keith   Bob Goddard
• [Asterisk-Users] lost g729 lic   Hermann Wecke
• [Asterisk-Users] VOIP-INFO   Michiel van Baak
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Jay Milk
• [Asterisk-Users] Manager API timestamps of events   Obelix
• [Asterisk-Users] Why does my name not show in the from address   Obelix
• [Asterisk-Users] VOIP-INFO   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Why does my name not show in the from address   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization   Steve Hanselman
• [Asterisk-Users] Asterisk to Cisco Unity   Steve Hanselman
• [Asterisk-Users] VOIP-INFO   Rob Thomas
• [Asterisk-Users] VOIP-INFO   Olle E. Johansson
• [Asterisk-Users] what is asteriskathome-1.0.iso?   Dean Collins
• [Asterisk-Users] VOIP-INFO   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Help! Zap echo on bridged calls   aturntablist
• [Asterisk-Users] Best platform   Serge Schumacher
• [Asterisk-Users] Best platform   Michiel van Baak
• [Asterisk-Users] Best platform   Mike M
• [Asterisk-Users] Best platform   Michiel van Baak
• [Asterisk-Users] Best platform   trixter http://www.0xdecafbad.com
• [Asterisk-Users] ATTN: Keith   Tracy Phillips
• [Asterisk-Users] Best platform   Paul
• [Asterisk-Users] ASTCC what has been changed   Ronald Wiplinger
• [Asterisk-Users] Voice quality of Softphones vs. IP Phones and Gateways.   Cenk Yabas
• [Asterisk-Users] ATTN: Keith   Dean Collins
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Matt
• [Asterisk-Users] Asterisk Users & Developers on their way to Madrid - Meet us there!   Olle E. Johansson
• [Asterisk-Users] Best platform   Mike M
• [Asterisk-Users] ASTCC what has been changed   Darren Wiebe
• [Asterisk-Users] ASTCC what has been changed   Ronald Wiplinger
• [Asterisk-Users] No path to translate from SIP/615-25c8(256) to SIP/601-27b6(4)   Ronald Wiplinger
• [Asterisk-Users] Best platform   Andrew Kohlsmith
• [Asterisk-Users] ATTN: Keith   Andrew Kohlsmith
• [Asterisk-Users] In Dial Application, reading the L(x[:y][:z]) parameter from database.   Cenk Yabas
• [Asterisk-Users] VoicePulse DTMF Problems Anyone?   Michael Stearne
• [Asterisk-Users] VoicePulse DTMF Problems Anyone?   Rich Adamson
• [Asterisk-Users] ATTN: Keith   Tracy Phillips
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Rich Adamson
• [Asterisk-Users] No path to translate from SIP/615-25c8(256) toSIP/601-27b6(4)   Joshua Colp
• [Asterisk-Users] Deleting Unavail Message   Michael Stearne
• [Asterisk-Users] Caller ID transforms   Pat Jensen
• [Asterisk-Users] Cisco 7960 mic generating noise on other end   Greg Oliver
• [Asterisk-Users] ATTN: Keith   Andrew Kohlsmith
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Esben Stien
• [Asterisk-Users] Manager API timestamps of events   Moises Silva
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Esben Stien
• [Asterisk-Users] No path to translate from SIP/615-25c8(256) toSIP/601-27b6(4)   Ronald Wiplinger
• [Asterisk-Users] Newie Questions   Esben Stien
• [Asterisk-Users] No path to translate from SIP/615-25c8(256)toSIP/601-27b6(4)   Joshua Colp
• [Asterisk-Users] ATTN: Keith   Robert Webb
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Andrew Kohlsmith
• [Asterisk-Users] ztdummy/rtc   Kevin Bockman
• [Asterisk-Users] VoicePulse DTMF Problems Anyone?   Rich Adamson
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   trixter http://www.0xdecafbad.com
• [Asterisk-Users] VoicePulse DTMF Problems Anyone?   Michael Stearne
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 77   Nguyen Trung Tin
• [Asterisk-Users] No path to translate from SIP/615-25c8(256)toSIP/601-27b6(4)   Ronald Wiplinger
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Joseph
• [Asterisk-Users] SIP_HEADER example   Denis Galvão - iSolve
• [Asterisk-Users] No path to translatefrom SIP/615-25c8(256)toSIP/601-27b6(4)   Joshua Colp
• [Asterisk-Users] No path to translate from SIP/615-25c8(256) toSIP/601-27b6(4)   Rich Adamson
• [Asterisk-Users] Shorewall Configuration for Asterisk Box   Samy Antoun
• [Asterisk-Users] Voice quality of Softphones vs. IP Phones an d Gateways.   mattf
• [Asterisk-Users] how to limit simultaneous calls   Juan Pablo Abuyeres
• [Asterisk-Users] ATTN: Keith (way OT)   Jay Milk
• [Asterisk-Users] how to limit simultaneous calls   Brian Roy
• [Asterisk-Users] RE: ztdummy/rtc   Kevin Bockman
• [Asterisk-Users] RE: Voicemail and MS Exchange Synchronization   Iassen Hristov
• [Asterisk-Users] VoicePulse DTMF Problems Anyone?   Rich Adamson
• [Asterisk-Users] ATTN: Keith   Bob Goddard
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Aidan Van Dyk
• [Asterisk-Users] No path to translate from SIP/615-25c8(256)toSIP/601-27b6(4)   Rich Adamson
• [Asterisk-Users] how to limit simultaneous calls   Juan Pablo Abuyeres
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   trixter http://www.0xdecafbad.com
• [Asterisk-Users] AreskiCC Calling Problem   Junaid Uppal
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Daryll Strauss
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Daryll Strauss
• [Asterisk-Users] Re: ztdummy/rtc   Tony Mountifield
• [Asterisk-Users] RE: ztdummy/rtc   Rob Thomas
• [Asterisk-Users] Flash hook not going through SPA-2002   Todd A. Riker
• [Asterisk-Users] how to limit simultaneous calls   Steve Wolfe
• [Asterisk-Users] Transcoding GSM to G723.1   Ade Agbero
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Zoa
• [Asterisk-Users] SIP Connection Timing Out BroadVoice   Michael Stearne
• [Asterisk-Users] Wildly inaccurate CDR records   Obelix
• [Asterisk-Users] AreskiCC Calling Problem   David John Walsh
• [Asterisk-Users] Re: ztdummy/rtc   Kevin Bockman
• [Asterisk-Users] Problems with IAX Trunks   Waldo Rubinstein
• [Asterisk-Users] Re: ztdummy/rtc - staticy audio   Kevin Bockman
• [OT] Why not use both? WAS: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Bryce Chidester
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Andrew Kohlsmith
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Andrew Kohlsmith
• [Asterisk-Users] SIP-H.323 dial tone and busy tone problem.   Carlos Alberto Lara de Hoyos
• [Asterisk-Users] Wildly inaccurate CDR records   snacktime
• [Asterisk-Users] Re: ztdummy/rtc - staticy audio   Rob Thomas
• [Asterisk-Users] Help with denighing access to certain numbers by CallerID   James Bean
• [Asterisk-Users] how to limit simultaneous calls   Juan Pablo Abuyeres
• [Asterisk-Users] Help with denighing access to certain numbers byCallerID   Rob Thomas
• [Asterisk-Users] howto write CDRs on two mysql servers   Max W Blackmer Jr
• [Asterisk-Users] Help with Oh323   Federico Alves
• [Asterisk-Users] Opinions of Sphinx?   Brian Roy
• [Asterisk-Users] Help with Oh323   Rob Thomas
• [Asterisk-Users] VoicePulse DTMF Problems Anyone?   Umair Bari
• [Asterisk-Users] VoicePulse DTMF Problems Anyone?   Michael Stearne
• [Asterisk-Users] Help with denighing access to certain numbersbyCallerID   James Bean
• [Asterisk-Users] ISDN Sub-Address   jurczak
• [Asterisk-Users] ASTCC what has been changed   Darren Wiebe
• [Asterisk-Users] Re: Asterisk forking, Was: Digium Website Update: Asterisk Business Edition   Lee Howard
• [Asterisk-Users] Re:DMS-500 CID name not in CDR   JR Richardson
• [Asterisk-Users] LCDC Integration/bounty   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization   Race Vanderdecken
• [Asterisk-Users] Best platform   Mike M
• [Asterisk-Users] Opinions of Sphinx?   mattf
• [Asterisk-Users] Re: Asterisk@Home connecting through firewall/router   C. Hatton Humphrey
• [Asterisk-Users] VoicePulse DTMF Problems Anyone?   Rich Adamson
• [Asterisk-Users] Opinions of Sphinx?   Race Vanderdecken
• [Asterisk-Users] SIP Connection Timing Out BroadVoice   Rich Adamson
• [Asterisk-Users] SPA-2001 features on analog side   Michael Stahl
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Pedro
• [Asterisk-Users] ATTN: Keith   Mr. James W. Laferriere
• [Asterisk-Users] SIP Connection Timing Out BroadVoice   Michael Stearne
• [Asterisk-Users] PRI Trouble   Michael Di Martino
• [Asterisk-Users] ATTN: Keith   C F
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   C F
• [Asterisk-Users] howto write CDRs on two mysql servers   Ronald Wiplinger
• [Asterisk-Users] SIP Connection Timing Out BroadVoice   Michael Stearne
• [Asterisk-Users] Not answering inbound a line used for outbound   jerry at voiptower.com
• [Asterisk-Users] Re: wrong numbers message   Luis Diaz
• [Asterisk-Users] Best platform   Dinesh Nair
• [Asterisk-Users] Not answering inbound a line used for outbound   Dinesh Nair
• [Asterisk-Users] ATTN: Keith   Dave Cotton
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Best platform   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Load balancing for each protocol   Yves
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Kevin P. Fleming
• [Asterisk-Users] ISDN Sub-Address   Armin Schindler
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   snacktime
• [Asterisk-Users] ISDN Sub-Address   jurczak
• [Asterisk-Users] PRI trouble   Michael Di Martino
• [Asterisk-Users] Load balancing for each protocol   Matt Riddell
• [Asterisk-Users] English vs American voice files   Matt Riddell
• [Asterisk-Users] Load balancing for each protocol   Michiel van Baak
• [Asterisk-Users] Opinions of Sphinx?   mattf
• [Asterisk-Users] Load balancing for each protocol   Yves
• [Asterisk-Users] Changing Email Templates   Graham Pearson
• [Asterisk-Users] Latest CVS and app_rxfax   Steve Underwood
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Bob Goddard
• [Asterisk-Users] Not answering inbound a line used for outbound   Rich Adamson
• [Asterisk-Users] Opinions of Sphinx?   end1r
• [Asterisk-Users] PRI trouble   Michael Di Martino
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Mark Charlton
• [Asterisk-Users] PRI trouble   Michael Di Martino
• [Asterisk-Users] Best bet ... IAX vs SIP   mr. barker
• [Asterisk-Users] DID Issue   Mark Johnson
• [Asterisk-Users] IAXy Pulse/Flash timing   Normando Marcolongo
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Esben Stien
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Bob Goddard
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Sahil Gupta
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Not answering inbound a line used for outbound   Jerry
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 81   patty McHenry
• [Asterisk-Users] Not answering inbound a line used for outboun   Jerry
• [Asterisk-Users] DID Issue   Chris Coulthurst
• [Asterisk-Users] GSM -> ULAW sound conversion   Kevin Bockman
• [Asterisk-Users] RE: GSM -> ULAW sound conversion   Kevin Bockman
• [Asterisk-Users] Asterisk to Cisco Unity   Simone
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Brian Capouch
• [Asterisk-Users] Asterisk to Cisco Unity   Simone
• [Asterisk-Users] RE: GSM -> ULAW sound conversion   Michiel van Baak
• [Asterisk-Users] RE: GSM -> ULAW sound conversion   Michiel van Baak
• [Asterisk-Users] RE: GSM -> ULAW sound conversion   Kevin Bockman
• [Asterisk-Users] Changing Email Templates   Peter Bowyer
• [Asterisk-Users] Unable To Register a SIP phone ... Help Needed   SYED ADEEL ALI
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Jason Becker
• [Asterisk-Users] ATTN: Keith - Seriously OT   Mark Musone
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Esben Stien
• [Asterisk-Users] ATTN: Keith   C F
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Esben Stien
• [Asterisk-Users] DID Issue   Mark Johnson
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   trixter http://www.0xdecafbad.com
• [Asterisk-Users] how to tell   Todd A. Riker
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   trixter http://www.0xdecafbad.com
• [Asterisk-Users] DID Issue   Chris Coulthurst
• [Asterisk-Users] how to tell   Chris Coulthurst
• [Asterisk-Users] how to tell   Todd A. Riker
• [Asterisk-Users] ZAP channel (X100P) won't detect call waiting   Todd A. Riker
• [Asterisk-Users] PRI trouble   Ty Carter
• [Asterisk-Users] how to tell   Chris Coulthurst
• [Asterisk-Users] *66 auto redial emulation?   Chris Coulthurst
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Andrew Kohlsmith
• [Asterisk-Users] Best bet ... IAX vs SIP   Nir Simionovich
• [Asterisk-Users] Re: Asterisk forking, Was: Digium Website Update: Asterisk Business Edition   Andrew Kohlsmith
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Andrew Kohlsmith
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Andrew Kohlsmith
• [Asterisk-Users] PRI trouble   Andrew Kohlsmith
• [Asterisk-Users] Asterisk Community Meeting in Sydney Australia   Shamsul Arefin
• [Asterisk-Users] Best bet ... IAX vs SIP   Steve Wolfe
• [Asterisk-Users] help for conference   craz sead
• [Asterisk-Users] 4+ Port FXS Analog Device   Lyle Giese
• [Asterisk-Users] Asterisk to Cisco Voip System Unity   Andy Hamilton
• [Asterisk-Users] how to tell   Todd A. Riker
• [Asterisk-Users] ATTN: Keith - Seriously OT   Terry H. Gilsenan
• [Asterisk-Users] how to tell   Todd A. Riker
• [Asterisk-Users] how to tell   Todd A. Riker
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Steve Wolfe
• [Asterisk-Users] PRI trouble   Andrew Kohlsmith
• [Asterisk-Users] PRI trouble   Kris Boutilier
• [Asterisk-Users] phone rings but caller doest hear it   Armand Sulter
• [Asterisk-Users] how to tell   Rich Adamson
• [Asterisk-Users] how to tell   Rich Adamson
• [Asterisk-Users] RE: GSM -> ULAW sound conversion   Kevin Bockman
• [Asterisk-Users] Not answering inbound a line used for outboun   Rich Adamson
• [Asterisk-Users] Best bet ... IAX vs SIP   mr. barker
• [Asterisk-Users] PRI Trouble   Steve Totaro
• [Asterisk-Users] POLYCOM IP 500 Setup   Matthew T. O'Connor
• [Asterisk-Users] SIP Authentication   Stojan Sljivic - GDS
• [Asterisk-Users] Macro support in realtime   Damon Estep
• [Asterisk-Users] Phantom incoming calls on x100p   Remco Barende
• [Asterisk-Users] Need Help with pickup *8   Kib Eki
• [Asterisk-Users] GXP2000 and hint LED's   Peter Svensson
• [Asterisk-Users] Asterisk code   Ibrar Ahmed
• [Asterisk-Users] Asterisk installation error after CVS update   Gundemarie Scholz
• [Asterisk-Users] Problem with DTMF Relay and Oh323   Federico Alves
• [Asterisk-Users] Modprobe wctdm hang at command prompt   Stojan Sljivic - GDS
• [Asterisk-Users] Problem with DTMF Relay and Oh323   Asterisk
• [Asterisk-Users] 407 Proxy Authentication Required   Shahan Kalutanthri
• [Asterisk-Users] Asterisk code   Matt Riddell
• [Asterisk-Users] about timeouts   Andrea Riela
• [Asterisk-Users] MySQL: max realistic size of extensions table.   Cenk Yabas
• [Asterisk-Users] SNOM, Asterisk and Attended transfer (bug?)   Steve Davies
• [Asterisk-Users] Guidance , for which card to buy   John Joseph
• [Asterisk-Users] PRI trouble   Mike M
• [Asterisk-Users] Cepstral partnership with Digium   Anton Krall
• [Asterisk-Users] Oh323 and Caller ID missing   Federico Alves
• [Asterisk-Users] Re: POLYCOM IP 500 Setup   Noah Miller
• [Asterisk-Users] Re: POLYCOM IP 500 Setup   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Matt
• [Asterisk-Users] nativ bridging problem with ilbc!!   Atuc
• [Asterisk-Users] presence and video conference   Juraj Bednar
• [Asterisk-Users] SNOM, Asterisk and Attended transfer (bug?)   Damon Estep
• [Asterisk-Users] SIP Authentication   Rick Baranowski
• [Asterisk-Users] SIP Authentication   Race Vanderdecken
• [Asterisk-Users] presence and video conference   Dean Collins
• [Asterisk-Users] wiki server session limit?   Damon Estep
• [Asterisk-Users] Asterisk to Cisco Unity   Race Vanderdecken
• [Asterisk-Users] Asterisk code   Race Vanderdecken
• [Asterisk-Users] Components and suggestions for an asterisk server with 9 to 17 POTS.   Ken Dresdell
• [Asterisk-Users] wiki server session limit?   Race Vanderdecken
• [Asterisk-Users] Re: Asterisk forking, Was: Digium Website Update: Asterisk Business Edition   Lee Howard
• [Asterisk-Users] T1 multiplexer (or ?) for failover in large installation   Mike
• [Asterisk-Users] snom 190: dial tone without registration?   alan
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Wiley Siler
• [Asterisk-Users] T1 multiplexer (or ?) for failover in largeinstallation   Brian C. Fertig
• [Asterisk-Users] Wildly inaccurate CDR records   Moises Silva
• [Asterisk-Users] snom 190: dial tone without registration?   Gavin Hamill
• [Asterisk-Users] SIP-H.323 dial tone and busy tone problem.   Moises Silva
• [Asterisk-Users] T1 multiplexer (or ?) for failover in largeinstallation   Trey Scarborough
• [Asterisk-Users] Interfacing to an IAD   Corwin Nichols
• [Asterisk-Users] DNIS and DID seeking confirmation   John Millican
• [Asterisk-Users] ProSLIC 3210 version 2 is too old.   Hugh L. Johnson
• [Asterisk-Users] Asterisk code   Matt Riddell
• [Asterisk-Users] Sip Trunking   Johann
• [Asterisk-Users] 407 Proxy Authentication Required   aram
• [Asterisk-Users] Interfacing to an IAD   Wiley Siler
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Esben Stien
• [Asterisk-Users] Re: Asterisk forking, Was: Digium Website Update:Asterisk Business Edition   The VoIP Connection
• [Asterisk-Users] DNIS and DID seeking confirmation   jltaylor
• [Asterisk-Users] Asterisk code   steve szmidt
• [Asterisk-Users] Re: Asterisk forking, Was: Digium Website Update:Asterisk Business Edition   Andrew Kohlsmith
• [Asterisk-Users] T1 multiplexer (or ?) for failover in large installation   qrss
• [Asterisk-Users] snom 190: dial tone without registration?   alan
• [Asterisk-Users] More on the IAD connection   Corwin Nichols
• [Asterisk-Users] Cepstral partnership with Digium   William Suffill
• [Asterisk-Users] More on the IAD connection   Wiley Siler
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Business Edition   Esben Stien
• [Asterisk-Users] Zaptel modules   Bouchra Benyelloul
• [Asterisk-Users] Zaptel modules   Moises Silva
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Asterisk connecting remote villages in western Uganda   Mark Summer
• [Asterisk-Users] T1 multiplexer (or ?) for failover in large installation   Leon Sun
• [Asterisk-Users] Hiss patch   Scott England
• [Asterisk-Users] DID in AMP with 2+ incoming lines   Tomas Florian
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Robert Hajime Lanning
• [Asterisk-Users] Asterisk connecting remote villages in westernUganda   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Unable to support trunking .... without zaptel timing   Geoff Manning
• [Asterisk-Users] Re: Voicemail and MS Exchange Synchronizatio n   David Brodbeck
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   David Brodbeck
• [Asterisk-Users] Re: Asterisk forking, Was: Digium Website Up date:Asterisk Business Edition   David Brodbeck
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Busi ness Edition   David Brodbeck
• [Asterisk-Users] Components and suggestions for an asterisk server with 9 to 17 POTS.   Andrew Latham
• [Asterisk-Users] VOIP-INFO.ORG website bug   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] Re: Digium Website Update: Asterisk BusinessEdition   Terry H. Gilsenan
• [Asterisk-Users] Group/Broadcast Voicemail   Robert Goodyear
• [Asterisk-Users] Voicemail and MS Exchange Synchronization   Race Vanderdecken
• [Asterisk-Users] MCI vs. XO/Allegiance   Wiley Siler
• [Asterisk-Users] Asterisk Evening in Melbourne (again!)nextThursday   Paul Hales
• [Asterisk-Users] MCI vs. XO/Allegiance   David Coulson
• [Asterisk-Users] MCI vs. XO/Allegiance   Paul Traue, Jr.
• [Asterisk-Users] MCI vs. XO/Allegiance   Leon Sun
• [Asterisk-Users] snom 190: dial tone without registration?   Karl Brose
• [Asterisk-Users] MCI vs. XO/Allegiance   Wiley Siler
• [Asterisk-Users] MCI vs. XO/Allegiance   mattf
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Damon Estep
• [Asterisk-Users] problem with pf and asterisk   Frank Cases
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Marcelo Pacheco
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Wiley Siler
• [Asterisk-Users] MCI vs. XO/Allegiance   Ariel Batista
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   David Coulson
• [Asterisk-Users] MCI vs. XO/Allegiance   Wiley Siler
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Wiley Siler
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Leon Sun
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Nir Simionovich
• [Asterisk-Users] IAX Issues...   Paul Dracevich
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Nir Simionovich
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   David Coulson
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Nir Simionovich
• [Asterisk-Users] Reading about G729   Kumara Jayaweera
• [Asterisk-Users] Need help connecting two * pcs with *@home   db_nz at paradise.net.nz
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Damon Estep
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Damon Estep
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Dave Weis
• [Asterisk-Users] ztcfg server crash   Rod Bacon
• [Asterisk-Users] ztcfg server crash   Jason Walker
• [Asterisk-Users] ztcfg server crash   Rod Bacon
• [Asterisk-Users] SIP Authentication   Damon Estep
• [Asterisk-Users] problem with pf and asterisk   C. Hatton Humphrey
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Eric Rees
• [Asterisk-Users] Group/Broadcast Voicemail   Chris Stinson
• [Asterisk-Users] Group/Broadcast Voicemail   Chris Stinson
• [Asterisk-Users] Need help connecting two * pcs with *@home   Andy Hamilton
• [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   Maron Kristófersson
• [Asterisk-Users] Asterisk connecting remote villages in western Uganda   support at sjobeck.com
• [Asterisk-Users] Re: Asterisk forking, Was: Digium Website Update:Asterisk Business Edition   support at sjobeck.com
• [Asterisk-Users] ztcfg server crash   Jason Walker
• [Asterisk-Users] octtel SP 4220 gateway and Asterisk   Eric Bishop
• [Asterisk-Users] SIP Listen to multiple ports   Prepaid
• [Asterisk-Users] SIP Listen to multiple ports   Damon Estep
• [Asterisk-Users] SIP Listen to multiple ports   Prepaid
• [Asterisk-Users] SIP Listen to multiple ports   Prepaid
• [Asterisk-Users] SIP Listen to multiple ports   Prepaid
• [Asterisk-Users] SIP Listen to multiple ports   Prepaid
• [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB   Irakli Natsvlishvili
• [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB   Damon Estep
• [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB   Damon Estep
• [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB   Shamsul Arefin
• [Asterisk-Users] SIP Listen to multiple ports   Shamsul Arefin
• [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB   Irakli Natsvlishvili
• [Asterisk-Users] SIP Listen to multiple ports   squid at cancun.com.mx
• [Asterisk-Users] Adtran TA 750 FXO Groundstart Mode   Syed Akbar
• [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB   Damon Estep
• [Asterisk-Users] Asterisk connecting remote villages in western Uganda   Caleb
• [Asterisk-Users] SIP-H.323 dial tone and busy tone problem.   Damian Minkov
• [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB   Irakli Natsvlishvili
• [Asterisk-Users] problem with pf and asterisk   Michiel van Baak
• [Asterisk-Users] SIP Listen to multiple ports   Damon Estep
• [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB   Damon Estep
• [Asterisk-Users] SIP Listen to multiple ports   Damon Estep
• [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB   Irakli Natsvlishvili
• [Asterisk-Users] SIP Listen to multiple ports   Juan J. Sierralta P.
• [Asterisk-Users] Keeping users, extensions, voicemail and so on in DB   Damon Estep
• [Asterisk-Users] SIP Listen to multiple ports   Damon Estep
• [Asterisk-Users] No mans problem?   Kib Eki
• [Asterisk-Users] SIP Listen to multiple ports   Juan J. Sierralta P.
• [Asterisk-Users] SIP Authentication   Stojan Sljivic - GDS
• [Asterisk-Users] SIP Authentication   Olle E. Johansson
• [Asterisk-Users] SIP Authentication   Olle E. Johansson
• [Asterisk-Users] SIP Authentication   Stojan Sljivic - GDS
• [Asterisk-Users] ztcfg server crash   Sergio Serrano
• [Asterisk-Users] Is there a problem when we want to transfer an incoming call to an external phone number   David Masure
• [Asterisk-Users] Call inband progress indication and zaphfc   Klaus-Peter Junghanns
• [Asterisk-Users] Max Retries Exceeded - IAX2. Network problem?   Paul Redstone
• [Asterisk-Users] Static CLID   IM.King
• [Asterisk-Users] AVAYA & Asteris & H323 chanel   Bohuslav Coufal
• [Asterisk-Users] ERROR[5539]: chan_zap.c:9750 setup_zap: Unable to load config zapata.conf   Yousef Herzallah
• [Asterisk-Users] ERROR[5539]: chan_zap.c:9750 setup_zap: Unable to load config zapata.conf   Matteo Brancaleoni
• [Asterisk-Users] How to connect to LVDX / opentelecoms.org   Roger Schreiter
• [Asterisk-Users] ERROR[5539]: chan_zap.c:9750 setup_zap: Unable to load config zapata.conf   Matteo Brancaleoni
• [Asterisk-Users] SIP to ZAP Dialplan   Walid Azab
• [Asterisk-Users] Is there a problem when we want to transfer an incoming call to an external phone number   Bob Goddard
• [Asterisk-Users] Is there a problem when we want to transfer an incoming call to an external phone number   Bob Goddard
• [Asterisk-Users] RTP Forwarding   David F. Bakker
• [Asterisk-Users] about timeouts   Andrea Riela
• [Asterisk-Users] Portable USB headset for VoIP   Paul Mahler
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 93   Nevruz Mesut Sahin
• [Asterisk-Users] Portable USB headset for VoIP   Francesco Peeters
• [Asterisk-Users] Is there a problem when we want to transfer anincoming call to an external phone number   David Masure
• [Asterisk-Users] Portable USB headset for VoIP   VoIP Newbie
• [Asterisk-Users] ERROR[6504]: chan_zap.c:6710 mkintf: Channel 24 is reserved for D-channel.   Yousef Herzallah
• [Asterisk-Users] Features.conf for secretary function   Alexis FECOURT
• [Asterisk-Users] RTP Forwarding   Florian Overkamp
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 93   Erdem HAKİ
• [Asterisk-Users] Best platform   Tzafrir Cohen
• [Asterisk-Users] Features.conf for secretary function   Michiel van Baak
• [Asterisk-Users] # no longer working   Michiel van Baak
• [Asterisk-Users] # no longer working   Michiel van Baak
• [Asterisk-Users] Long time to detect hang-up   Stojan Sljivic - GDS
• [Asterisk-Users] Caller ID   Stojan Sljivic - GDS
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Andrew Kohlsmith
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Andrew Kohlsmith
• [Asterisk-Users] Asterisk and Panasonic KX-TD1232   Amund Nygaard
• [Asterisk-Users] SIP Authentication   Stojan Sljivic - GDS
• [Asterisk-Users] How to setup a test number to know my extension number   Ronald Wiplinger
• [Asterisk-Users] problem with pf and asterisk   steve szmidt
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Rich Adamson
• [Asterisk-Users] HT-488 vs. SPA-3000?   Wai-Sun Chia
• [Asterisk-Users] How to setup a test number to know my extension number   Bryce Chidester
• [Asterisk-Users] How to setup a test number to know my extension number   Bob Goddard
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Nir Simionovich
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Wiley Siler
• [Asterisk-Users] SIP_HEADER - anybody using it?   Denis Galvão - iSolve
• [Asterisk-Users] Re: How to setup a test number to know my extension number   Ben Buxton
• [Asterisk-Users] Best platform   Mike M
• [Asterisk-Users] MCI vs. XO/Allegiance   Michael Graves
• [Asterisk-Users] HT-488 vs. SPA-3000?   alan
• [Asterisk-Users] VOIP-INFO down?   Marcel van Kaam, Fonetica
• [Asterisk-Users] Features.conf for secretary function   Alexis F.
• [Asterisk-Users] MCI vs. XO/Allegiance   Wiley Siler
• [Asterisk-Users] VOIP-INFO down?   Wiley Siler
• [Asterisk-Users] 488 Not Acceptable Here   Nabeel Jafferali
• [Asterisk-Users] VOIP-INFO down?   Huddleston, Robert
• [Asterisk-Users] Portable USB headset for VoIP   Jian Hong GUAN
• [Asterisk-Users] SIP to ZAP Dialplan   Moises Silva
• [Asterisk-Users] SIP Listen to multiple ports   Moises Silva
• [Asterisk-Users] VOIP-INFO down?   Damon Estep
• [Asterisk-Users] SIP Authentication   Damon Estep
• [Asterisk-Users] VOIP-INFO down?   Damon Estep
• [Asterisk-Users] VOIP-INFO down?   John Bittner
• [Asterisk-Users] Transfers on PRI connected channel banks and legacy PBX's   Steve Hanselman
• [Asterisk-Users] How to setup a test number to know my extensionnumber   Henrik Zachrau
• [Asterisk-Users] Info on ACD in Asterisk   ajay.kanth at wipro.com
• [Asterisk-Users] RTP Forwarding   Matteo Brancaleoni
• [Asterisk-Users] Asterisk and Panasonic KX-TD1232   Peter Svensson
• [Asterisk-Users] SIP_HEADER - anybody using it?   Charles Wang
• [Asterisk-Users] RTP Forwarding   David F. Bakker
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Matt
• [Asterisk-Users] Questions about contexts   Jerry
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Dean Collins
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Bryce Chidester
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Matt
• [Asterisk-Users] SIP Listen to multiple ports   Chris HARIGA
• [Asterisk-Users] VOIP-INFO down?   Bryce Chidester
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Pedro
• [Asterisk-Users] Re: Making Asterisk NOT Pickup a Line when Ringing?   Tony Mountifield
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Matt
• [Asterisk-Users] Adtran TA 750 FXO Groundstart Mode   Syed Akbar
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Wiley Siler
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Filippo Carone
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Mojo with Horan & Company, LLC
• [Asterisk-Users] -HEAD/--STABLE using 100% cpu   Kevin Bockman
• [Asterisk-Users] VOIP-INFO down?   Tim Pushor
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Brian McSpadden
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Andrew Kohlsmith
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Jerry
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Nathan Pralle
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Michael D Schelin
• [Asterisk-Users] Adtran TA 750 FXO Groundstart Mode   Huddleston, Robert
• [Asterisk-Users] Digium PCI-e Cards   Aaron Daniel
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Andrew Kohlsmith
• [Asterisk-Users] SIP_HEADER - anybody using it?   Denis Galvão - iSolve
• [Asterisk-Users] Portable USB headset for VoIP   Rusty Shackleford
• [Asterisk-Users] Adtran TA 750 FXO Groundstart Mode   Henry Coleman
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Mark Brown
• [Asterisk-Users] Zaptel problem on BSD   Mark Willis
• [Asterisk-Users] outgoing prefix dial plan   aryder at hollisinteractive.com
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Matt
• [Asterisk-Users] VOIP-INFO down?   Matt
• [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help me please   Kumara Jayaweera
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Brian McSpadden
• [Asterisk-Users] Questions about contexts   Moises Silva
• [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help meplease   Alexander Lopez
• [Asterisk-Users] VOIP-INFO down?   Brian C. Fertig
• [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help me please   Kevin Bockman
• [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card he lp me please   Colin Anderson
• [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help meplease   Marcel van Kaam, Fonetica
• [Asterisk-Users] Call parking in multi user environment   snacktime
• [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help me please   John Millican
• [Asterisk-Users] VOIP-INFO down?   Alexander Lopez
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Time Bandit
• [Asterisk-Users] Cannot handle frames in 2 format   Matthew Boehm
• [Asterisk-Users] -HEAD/--STABLE using 100% cpu   snacktime
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Andrew Kohlsmith
• [Asterisk-Users] -HEAD/--STABLE using 100% cpu   Waldo Rubinstein
• [Asterisk-Users] Zap Channels   Wiley Siler
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Bill McLaughlin
• [Asterisk-Users] Zap Channels   Tim Pushor
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Wiley Siler
• [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help meplease   Trey Scarborough
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Francesco Peeters
• [Asterisk-Users] -HEAD/--STABLE using 100% cpu   snacktime
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Huddleston, Robert
• [Asterisk-Users] Has anyone made a purchase from http://www.telephonyware.com/?   Ralph Frederick
• [Asterisk-Users] chan_sip.c: Maximum retries exceeded on call ........ for seqno 1 01 (Non-critical Response)   Geoff Manning
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Bryce Chidester
• [Asterisk-Users] how to remove + in CIDNumber   Asterisk guy
• [Asterisk-Users] VOIP-INFO down?   Tom Rymes
• AW: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   John van Oppen
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Wiley Siler
• [OT] Re: [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition   William Waites
• [Asterisk-Users] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU   luis.kibe at kddi.com.br
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Time Bandit
• [Asterisk-Users] Call being answered, but no audio on either end (Intermittent)   Geoff Manning
• [Asterisk-Users] ztcfg server crash   Rod Bacon
• [Asterisk-Users] E1 and SS7   Juan J. Sierralta P.
• [Asterisk-Users] OH323 Packetization   Rod Bacon
• [Asterisk-Users] RE: Asterisk-Users] te405p and dell poweredge   Ben Hencke
• [Asterisk-Users] te405p and dell poweredge   Ben Hencke
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Barton Fisher
• [Asterisk-Users] Digit Map for IP500 - prepend digits from phone   asterisk at txpe.net
• [Asterisk-Users] OH323 Packetization   Rod Bacon
• [Asterisk-Users] ATA186 & X100P - detect hangup   asterisk at txpe.net
• [Asterisk-Users] -HEAD/--STABLE using 100% cpu   Kevin Bockman
• [Asterisk-Users] OH323 Packetization   Rod Bacon
• [Asterisk-Users] Re: Making Asterisk NOT Pickup a Line when Ringing?   Rich Adamson
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Rich Adamson
• [Asterisk-Users] 488 Not Acceptable Here   Andres
• [Asterisk-Users] RJ45 instead RJ11 in Digium's TDM20B card help meplease   Chris Mason (Lists)
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Damon Estep
• [Asterisk-Users] Asterisk and grandstream weird call probs   Jamie Carl
• [Asterisk-Users] VOIP-INFO down?   Lboyd.Lists at fractionit
• [Asterisk-Users] Sipura SPA-841   Edwin Lam
• [Asterisk-Users] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU   luis.kibe at kddi.com.br
• [Asterisk-Users] snom 190: dial tone without registration?   Shaun Dwyer
• [Asterisk-Users] dialplan appdata separators   snacktime
• [Asterisk-Users] Snom hardware quality   Maron Kristófersson
• [Asterisk-Users] canreinvite=yes not working with sipura device.   Luki
• [Asterisk-Users] Sipura SPA-841   Tim Pushor
• [Asterisk-Users] Asterisk and grandstream weird call probs   Kevin Bockman
• [Asterisk-Users] how to make a dialplan on bases of Caller   Kamran Ahmad
• [Asterisk-Users] dialogic d/300pci-e1-120 compatible ?   craz sead
• [Asterisk-Users] RE: Call being answered, but no audio on either end   Gene Willingham
• [Asterisk-Users] Asterisk@Home and intermittent ring sound on callers end   Christopher Kenna
• [Asterisk-Users] how to make a dialplan on bases of Caller   Jamie Carl
• [Asterisk-Users] canreinvite=yes not working with sipura device.   Jon Gabrielson
• [Asterisk-Users] [PRI] TE110P   Michael L. Young
• [Asterisk-Users] E1 and SS7   Chris HARIGA
• [Asterisk-Users] Calling on all Polycom Experts   Ryan Stark
• [Asterisk-Users] Newbie question about pressing a key to be connected to the caller   Jason
• [Asterisk-Users] [PRI] TE110P   Timothy Costello
• [Asterisk-Users] [PRI] TE110P   Jason Walker
• [Asterisk-Users] Asterisk & outbound proxy?   Tim Pushor
• [Asterisk-Users] Sipura SPA-841   asterisk at txpe.net
• [Asterisk-Users] Comedian Mail Voicemail Goto Function   Kevin Kiely
• [Asterisk-Users] Questions about contexts   Rich Adamson
• [Asterisk-Users] upgrade problems...   Francois Meehan
• [Asterisk-Users] How to allow multiple codecs in A@H   stevanus
• [Asterisk-Users] outgoing prefix dial plan   Rich Adamson
• [Asterisk-Users] Newbie question about pressing a key to be connected to the caller   Jon Gabrielson
• [Asterisk-Users] Sipura SPA-841   Trevor Peirce
• [Asterisk-Users] How to allow multiple codecs in A@H   Mark Brown
• [Asterisk-Users] localize ${VM_DATE} ?   Louis-David Mitterrand
• SV: [Asterisk-Users] Asterisk and Panasonic KX-TD1232   Amund Nygaard
• [Asterisk-Users] TAPI for Dos (xbase / Clipper)   Remco Barende
• [Asterisk-Users] newbie question..   Sukardi Shahdan
• [Asterisk-Users] SIP REFER method.   Marcin Okraszewszki
• [Asterisk-Users] Old but Gold   support at sigmasdi.com
• [Asterisk-Users] Portable USB headset for VoIP   VoIP Newbie
• [Asterisk-Users] How to restrict access to * for a specific soft/hard phone model?   Jacek
• [Asterisk-Users] Calling on all Polycom Experts   Chris Mason (Lists)
• [Asterisk-Users] Stable Versions   Joseph
• [Asterisk-Users] app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)   Sandy Thomson
• [Asterisk-Users] Personalised Unavail / busy messages no longer play   Asterisk
• [Asterisk-Users] Dial more then 9 digits   Bohuslav Coufal
• [Asterisk-Users] How to restrict access to * for a specific soft/hard phone model?   Matteo Brancaleoni
• [Asterisk-Users] Sipura SPA 3000 FXO Setting India   Prepaid
• [Asterisk-Users] Dial more then 9 digits   altus
• [Asterisk-Users] Asterisk slow transferring calls   James Bean
• [Asterisk-Users] SIP call doesn't execute the 's'-extension   Tobias Wolf
• [Asterisk-Users] Dial more then 9 digits   Bohuslav Coufal
• [Asterisk-Users] Dial more then 9 digits   altus
• [Asterisk-Users] Dial more then 9 digits   Gavin Hamill
• [Asterisk-Users] Dial more then 9 digits   Bohuslav Coufal
• [Asterisk-Users] CDR's -> ODBC and logging IP's   niels at wxn.nl
• [Asterisk-Users] empty HDLC frame or bad CRC received   Bart Seresia
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Pedro
• [Asterisk-Users] Sipura SPA 3000 FXO Setting India   Prepaid
• [Asterisk-Users] Nasty little incident ...   Asterisk
• [Asterisk-Users] asterisk security   Georges Henroteaux
• [Asterisk-Users] Load problems   Calin Serbanescu
• [Asterisk-Users] Sipura SPA 3000 FXO Setting India   Rich Adamson
• [Asterisk-Users] RE: Calling on all Polycom Experts   David Gomillion
• [Asterisk-Users] Dial more then 9 digits   Rich Adamson
• [Asterisk-Users] Port Inquiry   Graham Pearson
• [Asterisk-Users] SIP call doesn't execute the 's'-extension   Rich Adamson
• [Asterisk-Users] Asterisk and Panasonic KX-TD1232   Michael J. Tubby B.Sc (Hons) G8TIC
• [Asterisk-Users] Old but Gold   Andrew Kohlsmith
• [Asterisk-Users] [Help] ZT_CHANCONFIG failed on channel 25   Yousef Herzallah
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Andrew Kohlsmith
• [Asterisk-Users] 488 Not Acceptable Here   Nabeel Jafferali
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Andrew Kohlsmith
• [Asterisk-Users] Caller ID   Juan Manuel Coronado Z.
• [Asterisk-Users] AVAYA & Asteris & H323 chanel   BJ Weschke
• [Asterisk-Users] Broadvoice and Inbound DTMF   Adam Robins
• [Asterisk-Users] newbie question..   Rich Adamson
• [Asterisk-Users] RE: Call being answered, but no audio on either end   Geoff Manning
• [Asterisk-Users] Sipura SPA 3000 FXO Setting India   Rich Adamson
• [Asterisk-Users] Nasty little incident ...   Rich Adamson
• [Asterisk-Users] asterisk security   Rich Adamson
• [Asterisk-Users] HT-488 vs. SPA-3000?   Rich Adamson
• [Asterisk-Users] Caller ID   Stojan Sljivic - GDS
• [Asterisk-Users] Broadvoice and Inbound DTMF   Adam Robins
• [Asterisk-Users] Nasty little incident ...   Asterisk
• [Asterisk-Users] AVAYA & Asteris & H323 chanel   Bohuslav Coufal
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Andrew Kohlsmith
• [Asterisk-Users] SIP call doesn't execute the 's'-extension   Tobias Wolf
• [Asterisk-Users] Asterisk and Max TNT   Michael Baird
• [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?   Rich Adamson
• [Asterisk-Users] user web interface   Daniel Poulsen
• [Asterisk-Users] SIP call doesn't execute the 's'-extension   Chris Stinson
• [Asterisk-Users] Handling -1 in dialplans   Thomas Andrews
• [Asterisk-Users] Error installing Asterisk with zaptel and libpri   Kumara Jayaweera
• [Asterisk-Users] app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)   Sandy Thomson
• [Asterisk-Users] [Help] ZT_CHANCONFIG failed on channel 25   Denis Galvão - iSolve
• [Asterisk-Users] HT-488 vs. SPA-3000?   Dan Littlejohn
• [Asterisk-Users] Nasty little incident ...   Rich Adamson
• [Asterisk-Users] SIP call doesn't execute the 's'-extension   Rich Adamson
• [Asterisk-Users] Asterisk SMS configuration   Jon Creasey
• [Asterisk-Users] asterisk gsm gateway hardware recommendation?   Juraj Bednar
• [Asterisk-Users] app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)   Sebastian Silva
• [Asterisk-Users] Asterisk SMS configuration   Tony Hoyle
• [Asterisk-Users] Port Inquiry   Dean Collins
• [Asterisk-Users] HT-488 vs. SPA-3000?   Michael Graves
• [Asterisk-Users] HT-488 vs. SPA-3000?   Tarpo, Louie
• [Asterisk-Users] Caller ID   Calin Serbanescu
• [Asterisk-Users] Asterisk SMS configuration   Jon Creasey
• [Asterisk-Users] app_dial.c:977 dial_exec_full: Unable to createchannel of type 'Zap' (cause 0)   Mehdi Chouikh
• [Asterisk-Users] app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)   Rich Adamson
• [Asterisk-Users] Question on cdr_odbc   Matt
• [Asterisk-Users] Re: Re: Digium Website Update: Asterisk Busi ness Edition   Esben Stien
• [Asterisk-Users] SIP transfer/REFER to voicemail problem   Bryan (JT) Ayers
• [Asterisk-Users] HT-488 vs. SPA-3000?   Tarpo, Louie
• [Asterisk-Users] Voip-info.org   Huddleston, Robert
• [Asterisk-Users] Help with Cron and Reload   Federico Alves
• [Asterisk-Users] Help with Cron and Reload   Neil Cherry
• [Asterisk-Users] Help with Cron and Reload   snacktime
• [Asterisk-Users] Unable to open master device '/dev/zap/ctl'   Kumara Jayaweera
• [Asterisk-Users] WiFi IP Phones   Anton Krall
• [Asterisk-Users] Voip-info.org   Robert Webb
• [Asterisk-Users] HT-488 vs. SPA-3000?   Rich Adamson
• [Asterisk-Users] phantom answer   support at sigmasdi.com
• [Asterisk-Users] Config files under CVS versioning system   Chris Earle (CBL)
• [Asterisk-Users] WiFi IP Phones   Cory Andrews
• [Asterisk-Users] Help with Cron and Reload   Moises Silva
• [Asterisk-Users] Unable to open master device '/dev/zap/ctl'   Sebastian Silva
• [Asterisk-Users] CellPhone BlueTooth adapater with Wireless Profile ??   TC
• [Asterisk-Users] Problem with overlap dialing   Nico Giefing
• [Asterisk-Users] E911 Interface   Chris
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 100   Jason
• [Asterisk-Users] Help with Cron and Reload   JD Austin
• [Asterisk-Users] WiFi IP Phones   Dean Collins
• [Asterisk-Users] CellPhone BlueTooth adapater with Wireless Profile ??   Francesco Peeters
• [Asterisk-Users] HT-488 vs. SPA-3000?   Dan Littlejohn
• [Asterisk-Users] RE: Question on cdr_odbc   Jason Kawakami
• [Asterisk-Users] Asterisk and Max TNT   Leon Sun
• [Asterisk-Users] HT-488 vs. SPA-3000?   Tarpo, Louie
• [Asterisk-Users] Caller ID on TelaSIP SIP Channel   Doug Eubanks
• [Asterisk-Users] ztcfg server crash   C F
• [Asterisk-Users] Caller ID on TelaSIP SIP Channel   Nico Giefing
• [Asterisk-Users] Nasty little incident ...   Steve Hanselman
• [Asterisk-Users] HT-488 vs. SPA-3000?   Brian Roy
• [Asterisk-Users] Caller ID on TelaSIP SIP Channel   Doug Eubanks
• [Asterisk-Users] HT-488 vs. SPA-3000?   Ariel Batista
• [Asterisk-Users] Sipura 841 Ringtones   Hugh L. Johnson
• [Asterisk-Users] WiFi IP Phones   Anton Krall
• [Asterisk-Users] WiFi IP Phones   Anton Krall
• [Asterisk-Users] Caller ID on TelaSIP SIP Channel   Nico Giefing
• [Asterisk-Users] Dundi - Multiple Results   Greg Boehnlein
• [Asterisk-Users] Sipura 841 Ringtones   Anton Krall
• [Asterisk-Users] Changing caller ID on a Zap channel   Jeff R Glassman
• [Asterisk-Users] WiFi IP Phones   Dean Collins
• [Asterisk-Users] WiFi IP Phones   Kanuri, Seshu (Company IT)
• [Asterisk-Users] echo cancellation on an iax2 channel   Script Head
• [Asterisk-Users] SIP to ZAP dialout without pre"0" (Asterisk <-> HiCom)   Michael Hamann
• [Asterisk-Users] HT-488 vs. SPA-3000?   Rich Adamson
• [Asterisk-Users] echo cancellation on an iax2 channel   Steve Underwood
• [Asterisk-Users] Problem with overlap dialing   Moises Silva
• [Asterisk-Users] zaptel witch smp   Wendell Almeida Silva
• [Asterisk-Users] Unable to open master device '/dev/zap/ctl'   Shamsul Arefin
• [Asterisk-Users] Grandstream ATA Toasted   Rod Bacon
• [Asterisk-Users] Nasty little incident ...   Jamie Carl
• [Asterisk-Users] WiFi IP Phones   Terry H. Gilsenan
• [Asterisk-Users] WiFi IP Phones   Terry H. Gilsenan
• [Asterisk-Users] WiFi IP Phones   Terry H. Gilsenan
• [Asterisk-Users] WiFi IP Phones   Nabeel Jafferali
• [Asterisk-Users] 1-800 DID in Alberta   Joseph
• [Asterisk-Users] WiFi IP Phones   Terry H. Gilsenan
• [Asterisk-Users] Includes include the includes?   Chris Mason (Lists)
• [Asterisk-Users] Includes include the includes?   Tarpo, Louie
• [Asterisk-Users] 1-800 DID in Alberta   Leon Sun
• [Asterisk-Users] Asterisk 1.0.7 Compile on Mac OS X Tiger 10.4.1   Darren Ellis
• [Asterisk-Users] This mailing list is being spam filtered on my site.   Gary Guthary
• [Asterisk-Users] octtel SP 4220 gateway and Asterisk   Eric Bishop
• [Asterisk-Users] Gnet Phones   Sean Cook
• [Asterisk-Users] Strange Inbound Ring Handling   Paul Dugas
• [Asterisk-Users] Problem with slin   Francois Meehan
• [Asterisk-Users] Incoming call stops at random with Teliax   Tracy Phillips
• [Asterisk-Users] SIP to PRI   Equipe du Royaume
• [Asterisk-Users] SIP to PRI   Jason Walker
• [Asterisk-Users] Bill seconds   Americo Sanchez C.
• [Asterisk-Users] octtel SP 4220 gateway and Asterisk   Eric Bishop
• [Asterisk-Users] VoiceXML? question   dave cantera
• [Asterisk-Users] Bridged-appearances   Tim Connolly
• [Asterisk-Users] SIP to PRI   Matthew Crocker
• [Asterisk-Users] Asterisk 1.0.7 Compile on Mac OS X Tiger 10.4.1   Michael Stearne
• [Asterisk-Users] terminating DID to FWD   Joseph
• [Asterisk-Users] Strange Inbound Ring Handling   Tim Connolly
• [Asterisk-Users] Gnet Phones   Nabeel Jafferali
• [Asterisk-Users] terminating DID to FWD   Nabeel Jafferali
• [Asterisk-Users] terminating DID to FWD   Darren Wiebe
• [Asterisk-Users] VoiceXML? question   El Flynn
• [Asterisk-Users] WiFi IP Phones   Terry H. Gilsenan
• [Asterisk-Users] asterisk with eurovoice.ro?   Lance Grover
• [Asterisk-Users] iax2 can't listen on virtual interface   Lance Grover
• [Asterisk-Users] Bill seconds   Juan Luis Moyano
• [Asterisk-Users] Bill seconds   Darren Wiebe
• [Asterisk-Users] Asterisk code   Ed Greenberg
• [Asterisk-Users] Grandstream ATA Toasted   Luki
• [Asterisk-Users] Changing caller ID on a Zap channel   Bryce Chidester
• [Asterisk-Users] Nasty little incident ...   Asterisk
• [Asterisk-Users] This mailing list is being spam filtered on my site.   Francesco Peeters
• [Asterisk-Users] newbie question..   Sukardi Shahdan
• [Asterisk-Users] iax2 can't listen on virtual interface   Boris Bakchiev
• [Asterisk-Users] Cheap Asterisk FXO PCI cards   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] Nasty little incident ...   steve at daviesfam.org
• [Asterisk-Users] Asterisk Integration with an SBC-410 phone system   J Scott Pitman
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Armin Schindler
• [Asterisk-Users] How to stop Asterisk from changing the SDP?   Stian Selnes
• [Asterisk-Users] Cheap Asterisk FXO PCI cards   Wai-Sun Chia
• [Asterisk-Users] Re: iax2 can't listen on virtual interface   Tony Mountifield
• [Asterisk-Users] Asterisk Live! CF   Torgeir Berg
• [Asterisk-Users] SER with Asterisk Problem   Mohamed A. Gombolaty
• [Asterisk-Users] Asterisk on Fedora Cora 3   Biagio Meirone
• [Asterisk-Users] How to stop Asterisk from changing the SDP?   steve at daviesfam.org
• [Asterisk-Users] phantom answer   support at sigmasdi.com
• [Asterisk-Users] How to stop Asterisk from changing the SDP?   Stian Selnes
• [Asterisk-Users] Asterisk Problems with FXO Ground Start Trunks and DID Wink Start Trunks   Syed Akbar
• [Asterisk-Users] Help with Cron and Reload   Tzafrir Cohen
• [Asterisk-Users] Grandstream phones losing registration with server.   Mark Brown
• [Asterisk-Users] How to stop Asterisk from changing the SDP?   steve at daviesfam.org
• [Asterisk-Users] MeetMe ERROR "Unable to dup channel"   sylvain garcia
• [Asterisk-Users] Re: iax2 can't listen on virtual interface   Boris Bakchiev
• [Asterisk-Users] How to stop Asterisk from changing the SDP?   Stian Selnes
• [Asterisk-Users] Busy, differences between SIP and Zaptel(bristuff)   Thomas Dingermann
• [Asterisk-Users] Error on incoming calls   f.zamboni
• [Asterisk-Users] Features.conf Set Language   sylvain garcia
• [Asterisk-Users] MeetMe ERROR "Unable to dup channel"   bdz at mailshack.com
• [Asterisk-Users] Grandstream phones losing registration with server.   Bob Goddard
• [Asterisk-Users] chan_h323 context   IM.King
• [Asterisk-Users] This mailing list is being spam filtered on my site.   Andrew Kohlsmith
• [Asterisk-Users] Grandstream phones losing registration withserver.   Mark Brown
• [Asterisk-Users] Zaphfc unable to dial out   zakhooi at ilse.nl
• [Asterisk-Users] WiFi IP Phones   Anton Krall
• [Asterisk-Users] WiFi IP Phones   Anton Krall
• [Asterisk-Users] Do includes include the includes   Chris Mason (Lists)
• [Asterisk-Users] WiFi IP Phones   Anton Krall
• [Asterisk-Users] WiFi IP Phones   Andrew Kohlsmith
• [Asterisk-Users] This mailing list is being spam filtered on my site.   Francesco Peeters
• [Asterisk-Users] unamble to dialout to mobiles and others "special" numbers   Marco Parmeggiani
• [Asterisk-Users] unamble to dialout to mobiles and others "special" numbers   Matteo Brancaleoni
• [Asterisk-Users] How to dimension Asterisk - that is used solely as callback server - only sending untranscoded voice between two ISDN channels on PRI ?   Robert Rozman
• [Asterisk-Users] SER and Asterisk question   Mohamed A. Gombolaty
• [Asterisk-Users] Fall back dialing   Chris Mason (Lists)
• [Asterisk-Users] SER and Asterisk question   Yair Hakak
• [Asterisk-Users] SER and Asterisk question   Steve Blair
• [Asterisk-Users] EuroISDN Italy - quadbri - zaptel.conf - whatsettings work ?   Robert Rozman
• [Asterisk-Users] reload from dialplan   Asterisk
• [Asterisk-Users] SER and Asterisk question   Mohamed A. Gombolaty
• [Asterisk-Users] Nasty little incident ...   Rich Adamson
• [Asterisk-Users] unamble to dialout to mobiles and others "special" numbers   Marco Parmeggiani
• [Asterisk-Users] terminating DID to FWD   Michael Graves
• [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # to work during a call   Michael J. Tubby B.Sc (Hons) G8TIC
• [Asterisk-Users] Bill seconds   Race Vanderdecken
• [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # to work during a call   Andrew Latham
• [Asterisk-Users] WiFi IP Phones   Dean Collins
• [Asterisk-Users] reload from dialplan   Asterisk
• Subject: [Asterisk-Users] asterisk gsm gateway hardware   Patrick Lidstone (Personal E-mail)
• [Asterisk-Users] Nasty little incident ...   Rich Adamson
• [Asterisk-Users] 1-800 DID in Alberta   Colin Anderson
• [Asterisk-Users] Case studies for Asterisk Voicemail   Alistair Cunningham
• [Asterisk-Users] reload from dialplan   Rich Adamson
• [Asterisk-Users] VoiceXML? question   Dean Collins
• [Asterisk-Users] Problem with 2 digium cards   Antoine Courouble
• [Asterisk-Users] Asterisk and Max TNT   Alexander Lopez
• [Asterisk-Users] AgentCallBacklogin (logout continued...)   1 2
• [Asterisk-Users] reload from dialplan   Asterisk
• [Asterisk-Users] Nobody picked up in 30000 ms   Kumara Jayaweera
• [Asterisk-Users] Error when compiling in freeTDS support   Matt
• [Asterisk-Users] Bridged-appearances   Mark Phillips
• [Asterisk-Users] SER and Asterisk question   Andreas Sikkema
• [Asterisk-Users] Asterisk and 2 line MGCP phone   CM Rahman Jr.
• [Asterisk-Users] Re: Error when compiling in freeTDS support   Matt
• [Asterisk-Users] Re: [Asterisk-biz] Case studies for Asterisk Voicemail   William Waites
• [Asterisk-Users] misdn and call hangup problem   Kib Eki
• [Asterisk-Users] reload from dialplan   Rich Adamson
• [Asterisk-Users] Case studies for Asterisk Voicemail   Michael Stearne
• [Asterisk-Users] Problems with IAX Trunks   Waldo Rubinstein
• [Asterisk-Users] Nobody picked up in 30000 ms   Robert Goodyear
• [Asterisk-Users] Re: [Asterisk-biz] Case studies for Asterisk Voicemail   Alistair Cunningham
• [Asterisk-Users] Case studies for Asterisk Voicemail   Alistair Cunningham
• [Asterisk-Users] Case studies for Asterisk Voicemail   Alistair Cunningham
• [Asterisk-Users] Coding a telemarketing call blocker   Tore Hansen
• [Asterisk-Users] have asterisk box #2 pick up calls.   Thomas Miller
• [Asterisk-Users] Intelligent maximum channels solution?   Remco Barende
• [Asterisk-Users] Bill seconds   Americo Sanchez C.
• [Asterisk-Users] Bill seconds   David John Walsh
• [Asterisk-Users] How to get started, what do I need?   Jayson Smith
• [Asterisk-Users] Case studies for Asterisk Voicemail   Michael Stearne
• [Asterisk-Users] Bill seconds   Leon Sun
• [Asterisk-Users] Coding a telemarketing call blocker   Walt Reed
• [Asterisk-Users] Bill seconds   Leon Sun
• [Asterisk-Users] Asterisk and 2 line MGCP phone   Florian Overkamp
• [Asterisk-Users] Asterisk and 2 line MGCP phone   CM Rahman Jr.
• [Asterisk-Users] Nasty little incident ...   steve at daviesfam.org
• [Asterisk-Users] Case studies for Asterisk Voicemail   Bill McLaughlin
• [Asterisk-Users] iax2 registry - auto reconnect ?   Jim Duda
• [Asterisk-Users] Case studies for Asterisk Voicemail   Michael Stearne
• [Asterisk-Users] Case studies for Asterisk Voicemail   Brian Capouch
• [Asterisk-Users] Case studies for Asterisk Voicemail   Dean Collins
• [Asterisk-Users] faxdetect config issues   Greg Blakely
• [Asterisk-Users] iax2 registry - auto reconnect ?   Jim Duda
• [Asterisk-Users] #(transfer) no longer working   Florian Overkamp
• [Asterisk-Users] Dell PowerEdge SC420 interrupt issue   Kevin Kiely
• [Asterisk-Users] iax2 registry - auto reconnect ?   Jonathan
• [Asterisk-Users] Comprehensive Asterisk Load Testing   Matt
• [Asterisk-Users] Nasty little incident ...   Asterisk
• [Asterisk-Users] iax2 registry - auto reconnect ?   Rich Adamson
• [Asterisk-Users] Re: iax2 can't listen on virtual interface   Lance Grover
• [Asterisk-Users] Sipura 3000 help   Adrian A
• [Asterisk-Users] Case studies for Asterisk Voicemail   Bill McLaughlin
• [Asterisk-Users] IAX load balancing   Yves
• [Asterisk-Users] SIP connection   Pedro Diaz
• [Asterisk-Users] #(transfer) no longer working   Michiel van Baak
• [Asterisk-Users] @Home AMP call recording documentation   Dan Littlejohn
• [Asterisk-Users] Re: @Home AMP call recording documentation   Dan Littlejohn
• [Asterisk-Users] @Home AMP call recording documentation   Dean Collins
• [Asterisk-Users] Case studies for Asterisk Voicemail   Durf
• [Asterisk-Users] Console ALSA Sound   Dan Perik
• [Asterisk-Users] Console ALSA Sound   Sebastian Silva
• [Asterisk-Users] Console ALSA Sound   Bryce Chidester
• [Asterisk-Users] #(transfer) no longer working   Michiel van Baak
• [Asterisk-Users] MFC/R2   j_amorim
• [Asterisk-Users] Console ALSA Sound   Conrad Wood
• [Asterisk-Users] Sipura 3000 help   Luki
• [Asterisk-Users] Sipura 3000 help   Tarpo, Louie
• [Asterisk-Users] Sipura 3000 help   Adrian A
• [Asterisk-Users] Console ALSA Sound   Sean Kennedy
• [Asterisk-Users] Sipura 3000 help   Rich Adamson
• [Asterisk-Users] Multiple Sipura 3000   Martin Roy
• [Asterisk-Users] rxfax problem - libspandsp issue?   Roger Schreiter
• [Asterisk-Users] Bill seconds   C F
• [Asterisk-Users] Grandstream ATA Toasted   Rod Bacon
• [Asterisk-Users] faxdetect config issues   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] Call parking in multi user environment   C F
• [Asterisk-Users] Do includes include the includes   Chris Mason (Lists)
• [Asterisk-Users] meetme - conf-invalid   scott
• [Asterisk-Users] rxfax problem - libspandsp issue?   qrss
• [Asterisk-Users] meetme - conf-invalid   Kevin Bockman
• [Asterisk-Users] meetme - conf-invalid   qrss
• [Asterisk-Users] Bill seconds   Terry H. Gilsenan
• [Asterisk-Users] meetme - conf-invalid   Kevin Bockman
• [Asterisk-Users] Viva Madrid!   Nicolás Gudiño
• [Asterisk-Users] Newbie question about pressing a key to, be connected to the caller   Jason
• [Asterisk-Users] Call parking in multi user environment   snacktime
• [Asterisk-Users] Newbie question about pressing a key to, be connected to the caller   Kevin Bockman
• [Asterisk-Users] MFC/R2   Steve Underwood
• [Asterisk-Users] Bill seconds   Tony Hoyle
• [Asterisk-Users] rxfax problem - libspandsp issue?   Steve Underwood
• [Asterisk-Users] Re: @Home AMP call recording documentation   Jason Becker
• [Asterisk-Users] Bill seconds   Americo Sanchez C.
• [Asterisk-Users] Bill seconds   Americo Sanchez C.
• [Asterisk-Users] Case studies for Asterisk Voicemail   Bill McLaughlin
• [Asterisk-Users] Bill seconds   Bill McLaughlin
• [Asterisk-Users] Bill seconds   Race Vanderdecken
• [Asterisk-Users] Console ALSA Sound   Sean Kennedy
• [Asterisk-Users] Routing SIP to Cisco routers running IOS 12.3+   Bryan M. Johns
• [Asterisk-Users] iaxy and cvs head...   Francois Meehan
• [Asterisk-Users] Bill seconds   Darren Wiebe
• [Asterisk-Users] Dial Commands "D" Option Question   Nate Kapi
• [Asterisk-Users] Dial Commands "D" Option Question   Bryan M. Johns
• [Asterisk-Users] Problem with monitor.   Daniel Eriksson
• [Asterisk-Users] Unable to open master device '/dev/zap/ctl'   f6hqz-m at wanadoo.fr
• [Asterisk-Users] How to detect telco Message Waiting Indicator (WMI)   Brian Martin
• [Asterisk-Users] Re: Dell PowerEdge SC420 interrupt issue   Paul Redstone
• [Asterisk-Users] Re: Dell PowerEdge SC420 interrupt issue   Peter Svensson
• [Asterisk-Users] inbound agent recording filename   Asterisk
• [Asterisk-Users] Caller ID   Stojan Sljivic - GDS
• [Asterisk-Users] Re: meetme - conf-invalid   Tony Mountifield
• [Asterisk-Users] Re: meetme - conf-invalid   Tony Mountifield
• [Asterisk-Users] Junk at the beginning of frame   Asterisk
• [Asterisk-Users] Viva Madrid!   Wojciech Tryc
• [Asterisk-Users] Long time to detect hang-up   Stojan Sljivic - GDS
• [Asterisk-Users] ata186 IVR problem   Betül Gözlükoğlu
• [Asterisk-Users] Multiple Sipura 3000   Chris Stenton
• [Asterisk-Users] Dial timeout when server down   Yves
• [Asterisk-Users] EuroISDN Italy - quadbri - zaptel.conf - whatsettings work ?   Emanuele Pucciarelli
• [Asterisk-Users] Comprehensive Asterisk Load Testing   Zoa
• [Asterisk-Users] Dial timeout when server down   Kris Boutilier
• [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2   Manuel Casal
• [Asterisk-Users] Bill seconds   David John Walsh
• [Asterisk-Users] Sip INFO DTMF over satellite   lists at bazaar.co.ke
• [Asterisk-Users] Miax: Digital voice channel when connecting to asterisk   Roger Schreiter
• [Asterisk-Users] Includes include the includes?   Chris Mason (Lists)
• [Asterisk-Users] Call group channel limits   Iain Sims
• [Asterisk-Users] Dell PowerEdge + TDM   David Hajek
• [Asterisk-Users] Analog modems behind an Asterisk server?   Christian Schnell
• [Asterisk-Users] Analog modems behind an Asterisk server?   Florian Overkamp
• [Asterisk-Users] Dial Commands "D" Option Question   Rich Adamson
• [Asterisk-Users] Agents/Queues Contexts   Waldo Rubinstein
• [Asterisk-Users] Multiple Sipura 3000   Rich Adamson
• [Asterisk-Users] RE: Asterisk Google API applications - $4500 bounties available   Dean Collins
• [Asterisk-Users] Dell PowerEdge + TDM   David John Walsh
• [Asterisk-Users] Need Last App-Fax Source   Damian Minkov
• [Asterisk-Users] Dell PowerEdge + TDM   David Hajek
• [Asterisk-Users] Analog modems behind an Asterisk server?   Rich Adamson
• [Asterisk-Users] Strange problem with G711/G729, Cisco and Grandstream   Jason Williams
• [Asterisk-Users] Asterisk box as a billing machine in a PSTN network   Africa Digital
• [Asterisk-Users] Multiple Sipura 3000   Tim P
• [Asterisk-Users] auto-dial dial status   Marco Parmeggiani
• [Asterisk-Users] No ringing tone on outgoing SIP trunk   Normando Marcolongo
• [Asterisk-Users] Dell PowerEdge + TDM   Oswaldo Arratia
• [Asterisk-Users] Includes include the includes?   Tarpo, Louie
• [Asterisk-Users] WiFi IP Phones   Daryl G. Jurbala
• [Asterisk-Users] 2nd Dialtone after answer   Oswaldo Arratia
• [Asterisk-Users] WiFi IP Phones   Daryl G. Jurbala
• [Asterisk-Users] Calculating the lenght of time in a call queue?   Shad Mortazavi
• [Asterisk-Users] Calculating the lenght of time in a call queue?   Niall McCarthy
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Daryl G. Jurbala
• [Asterisk-Users] 2nd Dialtone after answer   Chris Coulthurst
• [Asterisk-Users] Programinng Aplication with Music on Hold   Jose Raul Pineda Lemus
• [Asterisk-Users] 2nd Dialtone after answer   David John Walsh
• [Asterisk-Users] WiFi IP Phones   trixter http://www.0xdecafbad.com
• [Asterisk-Users] MFC/R2   j_amorim
• [Asterisk-Users] Presence and IM?   Bjorn
• [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2   Marco Parmeggiani
• [Asterisk-Users] 2nd Dialtone after answer   Henry Coleman
• [Asterisk-Users] Presence and IM?   Dean Collins
• [Asterisk-Users] Can't switch span to E1-mode   Yousef Herzallah
• [Asterisk-Users] Bill seconds   C F
• [Asterisk-Users] Bill seconds   C F
• [Asterisk-Users] G729   David
• [Asterisk-Users] G729   Bruce Komito
• [Asterisk-Users] tdm400p not working after cvs-head update   David Romero
• [Asterisk-Users] SIP transfer/REFER to voicemail problem   B Ayers
• [Asterisk-Users] G729   Erick Weber V.
• [Asterisk-Users] Bill seconds   Jean-Michel Hiver
• [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2   Manuel Casal
• [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2   Marco Parmeggiani
• [Asterisk-Users] txfax 18Kb file problem   Vladyslav
• [Asterisk-Users] PIX Firewall Ports and Access-Lists   Geoff Manning
• [Asterisk-Users] Can't switch span to E1-mode   izo
• [Asterisk-Users] Dell PowerEdge + TDM   David Hajek
• [Asterisk-Users] RE:Calculating the lenght of time in a call queue?   Shad Mortazavi
• [Asterisk-Users] Phantom problem authenticating IAX2 with RSA   Jon Lewis
• SV: [Asterisk-Users] Presence and IM?   Bjorn
• [Asterisk-Users] PLEASE HELP X100P no responding   Christian Callejon
• [Asterisk-Users] Spandsp - fax problem   miguel
• SV: [Asterisk-Users] Presence and IM?   Bryan M. Johns
• [Asterisk-Users] Phone lookup   Aaron Daniel
• SV: [Asterisk-Users] Presence and IM?   Bryan M. Johns
• [Asterisk-Users] Multiple phones on a Zap FXS extension   Kelly Opal
• SV: [Asterisk-Users] Presence and IM?   Time Bandit
• [Asterisk-Users] G729   Carlos Chavez
• [Asterisk-Users] Re: tdm400p not working after cvs-head update   David Romero
• [Asterisk-Users] Console ALSA Sound   Conrad Beckert
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Robert Goodyear
• [Asterisk-Users] MGCP files for Polycom   Rick Baranowski
• [Asterisk-Users] Re: tdm400p not working after cvs-head update   Soner Tari
• [Asterisk-Users] Console ALSA Sound   Conrad Beckert
• [Asterisk-Users] callqueues confused :(   Neil Bullock
• [Asterisk-Users] Bill seconds   Terry H. Gilsenan
• [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?   Darren Wright
• [Asterisk-Users] Asterisk ael files   Shidan
• [Asterisk-Users] ASTCC Rate Calculation   Darren Wiebe
• SV: [Asterisk-Users] Presence and IM?   rsenykoff at harrislogic.com
• [Asterisk-Users] Queue/How to get the number of incoming calls   Gary Li
• [Asterisk-Users] Local numbers   jonr
• [Asterisk-Users] Unable to find a path from g729 to gsm   Kumara Jayaweera
• [Asterisk-Users] callqueues confused :(   Steve Totaro
• [Asterisk-Users] tdm400p not working after cvs-head update   Steve Totaro
• [Asterisk-Users] AreskiCC and asterisk   zhu
• [Asterisk-Users] allowing outside dialing only for SIP users   David Arendt
• [Asterisk-Users] Analog modems behind an Asterisk server?   Christian Schnell
• [Asterisk-Users] How to view all call detail in asterisk server   SUBHASH RAVADA
• [Asterisk-Users] Re: Asterisk ael files   Tony Mountifield
• [Asterisk-Users] Re: Analog modems behind an Asterisk server?   Christian Schnell
• [Asterisk-Users] allowing outside dialing only for SIP users   Wilson Pickett
• [Asterisk-Users] Local numbers   Wilson Pickett
• [Asterisk-Users] Re: Asterisk ael files   Wilson Pickett
• [Asterisk-Users] TTS   trixter http://www.0xdecafbad.com
• [Asterisk-Users] error compiling asterisk   Ronald Wiplinger
• [Asterisk-Users] error compiling asterisk   Dave Cotton
• [Asterisk-Users] error compiling asterisk   Ronald Wiplinger
• [Asterisk-Users] E1 to E1 connection   Cenk Yabas
• [Asterisk-Users] UK SMS Config problems   Jon Creasey
• [Asterisk-Users] ASTCC Rate Calculation   Ronald Wiplinger
• 回复： [Asterisk-Users] How to view all call detail in asterisk server   Gary Li
• [Asterisk-Users] Re: Analog modems behind an Asterisk server?   Rich Adamson
• [Asterisk-Users] allowing outside dialing only for SIP users   David Arendt
• [Asterisk-Users] Unable to find a path from g729 to gsm   Rich Adamson
• [Asterisk-Users] ASTCC Rate Calculation   Darren Wiebe
• [Asterisk-Users] FATAL: Error running install command for wctdm   Ronald Wiplinger
• [Asterisk-Users] FATAL: Error running install command for wctdm   Richard Lyman
• [Asterisk-Users] PIX Firewall Ports and Access-Lists   Geoff Manning
• [Asterisk-Users] FATAL: Error running install command for wctdm   Ronald Wiplinger
• [Asterisk-Users] ASTCC Rate Calculation   Rusty Shackleford
• [Asterisk-Users] FATAL: Error running install command for wctdm   Jason Becker
• [Asterisk-Users] Asterisk - fax - spandsp   Les Caroto
• [Asterisk-Users] 2nd Dialtone after answer   Oswaldo Arratia
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 68   Nguyen Trung Tin
• [Asterisk-Users] Unable to make outbound calls   Balaji NJL
• [Asterisk-Users] Want to test my * behind firewall   Balaji NJL
• [Asterisk-Users] IAX with shaw cable not going through   Joseph
• [Fwd: [Asterisk-Users] IAX with shaw cable not going through]   Joseph
• [Asterisk-Users] IAX with shaw cable not going through   Joseph
• [Asterisk-Users] H323 implementations   Richard Scobie
• [Asterisk-Users] ASTCC Rate Calculation   Darren Wiebe
• [Asterisk-Users] [partialy - solved] IAX with shaw cable not going through   Joseph
• [Asterisk-Users] IAX with shaw cable not going through   John Cianfarani
• [Asterisk-Users] [partialy - solved] IAX with shaw cable not going through   Robert Webb
• [Asterisk-Users] [partialy - solved] IAX with shaw cable not going through   Joseph
• [Asterisk-Users] ASTCC Rate Calculation   Rusty Shackleford
• [Asterisk-Users] ASTCC Rate Calculation   Darren Wiebe
• [Asterisk-Users] Three way calling with Cisco 12SP+   Blake OPS
• [Asterisk-Users] New Voip-info.org mirror/translation   Matt Darnell
• [Asterisk-Users] FATAL: Error running install command for wctdm   Ronald Wiplinger
• [Asterisk-Users] How to setup two Asterisk boxes - keeping the registration   Ronald Wiplinger
• [Asterisk-Users] channel.c:1884 set_format: Unable to find a path from g729 to gsm   Tim Pushor
• [Asterisk-Users] channel.c:1884 set_format: Unable to find a path from g729 to gsm   Kumara Jayaweera
• [Asterisk-Users] Bill seconds   C F
• [Asterisk-Users] Bill seconds   Terry H. Gilsenan
• [Asterisk-Users] Asterisk on vpbx exited on signal 11. Might want to take a peek.   Ronald Wiplinger
• [Asterisk-Users] Re: [Asterisk-Dev] chan_capi-cm-0.5 release announcement   Armin Schindler
• [Asterisk-Users] Is actually somebody doing it: load balancing 20 asterisk servers   Ronald Wiplinger
• [Asterisk-Users] Re: [Asterisk-Dev] chan_capi-cm-0.5 release announcement   Armin Schindler
• [Asterisk-Users] Now it is working   Kumara Jayaweera
• [Asterisk-Users] SER and Asterisk question   Mohamed A. Gombolaty
• [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2   Bob Goddard
• [Asterisk-Users] Can't switch span to E1-mode   Bob Goddard
• [Asterisk-Users] stale nonce received   Ronald Wiplinger
• [Asterisk-Users] Console ALSA Sound   Emanuele Pucciarelli
• [Asterisk-Users] Re: rxfax problem - libspandsp issue?   Roger Schreiter
• [Asterisk-Users] Libtiff 3.5.7 - recommended version for spandsp   Roger Schreiter
• [Asterisk-Users] bluetooth audio and asterisk   trixter http://www.0xdecafbad.com
• [Asterisk-Users] app_curl.so: Can't locate module sound-service-0-3   Mike M
• [Asterisk-Users] missing mysql cdr records   Rosario Pingaro
• [Asterisk-Users] I don't get a queue_log with my version of asterisk (0.7.1).   Shad Mortazavi
• [Asterisk-Users] bluetooth audio and asterisk   Dean Collins
• [Asterisk-Users] bluetooth audio and asterisk   Juraj Bednar
• [Asterisk-Users] asterisk and fayn.cz   Juraj Bednar
• SV: [Asterisk-Users] Presence and IM?   Juraj Bednar
• [Asterisk-Users] WiFi IP Phones   Juraj Bednar
• [Asterisk-Users] outgoing call routing   Jose Vicente Ortega
• [Asterisk-Users] WiFi IP Phones   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Bill seconds   C F
• [Asterisk-Users] chan_capi-cm-0.5.1 fixup release   Armin Schindler
• [Asterisk-Users] *67 with Sipura 3000   Martin Roy
• [Asterisk-Users] MySQL to static .conf   Juan Luis Moyano
• [Asterisk-Users] Problem with astperl primitives say... in astcc   Bruce Komito
• [Asterisk-Users] Zaptel and Zapata Conf's   Barton Fisher
• [Asterisk-Users] chanisavail...not workin with SIP and IAX   Dan Fernandez
• [Asterisk-Users] NAT/Proxy advise   Neil Bullock
• [Asterisk-Users] tos problem   Calin Serbanescu
• [Asterisk-Users] tos problem   Doug Lytle
• [Asterisk-Users] [SOLVED IAX with shaw cable not going through   Joseph
• [Asterisk-Users] Bill seconds   Terry H. Gilsenan
• [Asterisk-Users] Unable to make outbound calls   Balaji NJL
• [Asterisk-Users] Want to test my * behind firewall   Balaji NJL
• [Asterisk-Users] NAT/Proxy advise   Tony Hoyle
• [Asterisk-Users] Bill seconds   C F
• [Asterisk-Users] Panasonic KX-TD1232   Dan Morin
• [Asterisk-Users] Bill seconds [so far off topic that it has become a singularity]   Terry H. Gilsenan
• [Asterisk-Users] Panasonic KX-TD1232   Shane Young
• [Asterisk-Users] Polycom 500 Sound Problem   Matt
• [Asterisk-Users] bluetooth audio and asterisk   Jay Milk
• [Asterisk-Users] Bill seconds [so far off topic that it hasbecome a singularity]   Bill McLaughlin
• [Asterisk-Users] Scratchy audio on Bridged PRI Calls   Rod Bacon
• [Asterisk-Users] Premptible Linux Kernel   Rod Bacon
• [Asterisk-Users] bluetooth audio and asterisk   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Zaptel HEAD with * Stable?   Rod Bacon
• [Asterisk-Users] tos problem   Calin Serbanescu
• [Asterisk-Users] help for making several calls at the same time..   Erdem HAKİ
• [Asterisk-Users] No CDR Records   Andrew Tranter
• [Asterisk-Users] webvmail debian package   sylvain garcia
• [Asterisk-Users] Debian Vs Fedora   Syed Akbar
• [Asterisk-Users] Debian Vs Fedora   sylvain garcia
• [Asterisk-Users] Re: Console ALSA Sound   Louis-David Mitterrand
• [Asterisk-Users] Panasonic KX-TD1232   Peter Svensson
• [Asterisk-Users] OH323 with g723   Bashir Ullah - www.Lamsre.Com
• [Asterisk-Users] OH323 with g723   Erdem HAKİ
• [Asterisk-Users] sipredirect question   Axel Schemberg
• [Asterisk-Users] Re: [Asterisk-biz] sipredirect question   Emanuele Pucciarelli
• [Asterisk-Users] webvmail debian package   sylvain garcia
• [Asterisk-Users] $0-per-month (pay as you go) provider with T.38?   Adam Megacz
• [Asterisk-Users] oneTouchVoicemail issue with Polycom 1.5.2   Louis-David Mitterrand
• [Asterisk-Users] Libtiff 3.5.7 - recommended version for spandsp   Marco Parmeggiani
• [Asterisk-Users] CPU load 100% when SIP register   Asterisk
• [Asterisk-Users] AGI/PHP errors   Jeromy Grimmett
• [Asterisk-Users] call file ignored?   Remco Barende
• [Asterisk-Users] SIP Ad-Hoc Conferencing with Asterisk   Joshua Colp
• [Asterisk-Users] $0-per-month (pay as you go) provider with T.38?   Joshua Colp
• [Asterisk-Users] second isdn line doesn't work with avm c2 card   Simone Cittadini
• [Asterisk-Users] NAT/Proxy advise   Rich Adamson
• [Asterisk-Users] tos problem   Rich Adamson
• [Asterisk-Users] call file ignored?   jurczak
• [Asterisk-Users] AGI/PHP errors   Michiel van Baak
• [Asterisk-Users] AGI/PHP errors   Matthew Boehm
• [Asterisk-Users] MGCP and SIP clients   Mathias Röhl
• [Asterisk-Users] call file ignored?   Remco Barende
• [Asterisk-Users] call file ignored?   jurczak
• [Asterisk-Users] call file ignored?   Marco Parmeggiani
• [Asterisk-Users] bluetooth audio and asterisk   Jay Milk
• [Asterisk-Users] VoiceMail   Waldo Rubinstein
• [Asterisk-Users] IAX with shaw cable not going through   Colin Anderson
• [Asterisk-Users] IAX with shaw cable not going through   Colin Anderson
• [Asterisk-Users] app_valetparking.c   Nenad Radosavljevic
• [Asterisk-Users] chan_h323 vs chan_oh323 & chan_ooh323   Atif Rasheed
• [Asterisk-Users] VoiceMail   Greg Jones Media
• [Asterisk-Users] VoiceMail   sylvain garcia
• [Asterisk-Users] Libtiff 3.5.7 - recommended version for spandsp   Marco Parmeggiani
• [Asterisk-Users] VoiceMail   John Bittner
• [Asterisk-Users] call file ignored?   Remco Barende
• [Asterisk-Users] LookupCIDName on outgoing calls   Leandro Morgado
• [Asterisk-Users] TxFax: can't get a fax to destination (log inside)   Marco Parmeggiani
• [Asterisk-Users] VoiceMail   Nicolas FOURNIL
• [Asterisk-Users] Panasonic KX-TD1232   asterisk
• [Asterisk-Users] Zaptel HEAD with * Stable?   Chad Osmond
• [Asterisk-Users] outgoing call routing   Sergio Serrano
• [Asterisk-Users] Unable to make outbound calls   Time Bandit
• [Asterisk-Users] Can't get TDM04B to work!   Angus Comber
• [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # towork during a call   Michael J. Tubby B.Sc (Hons) G8TIC
• [Asterisk-Users] Zaptel HEAD with * Stable?   Florian Overkamp
• [Asterisk-Users] TxFax: can't get a fax to destination (log inside)   Lee Howard
• [Asterisk-Users] app_valetparking.c   Paul Zimm
• [Asterisk-Users] oneTouchVoicemail issue with Polycom 1.5.2   C F
• [Asterisk-Users] Bill seconds [so far off topic that it has become a singularity]   C F
• [Asterisk-Users] Polycom 500 Sound Problem   C F
• [Asterisk-Users] bluetooth audio and asterisk   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Zaptel HEAD with * Stable?   Doug Lytle
• [Asterisk-Users] bristuff-0.2.0-RC8g: zaptel error in suse 9.2   Tony Nichols
• [Asterisk-Users] Dell PowerEdge + TDM   Tony Nichols
• [Asterisk-Users] outgoing call routing   Andrew Latham
• [Asterisk-Users] Polycom 500 Sound Problem   Bryan M. Johns
• [Asterisk-Users] Re: app_valetparking.c for * STABLE (1.0.X)   Nenad Radosavljevic
• [Asterisk-Users] VoiceMail   Time Bandit
• [Asterisk-Users] bluetooth audio and asterisk   Jay Milk
• [Asterisk-Users] Zaptel HEAD with * Stable?   Seth Remington
• [Asterisk-Users] webvmail debian package   Scott Kamp
• [Asterisk-Users] ASTCC Rate Calculation   Rusty Shackleford
• [Asterisk-Users] AGI/PHP errors   Jay Milk
• [Asterisk-Users] Re: [Asterisk-Dev] chan_capi-cm-0.5 release announcement   Armin Schindler
• [Asterisk-Users] What should I get for SOHO TDM card or sipura3000?   Robert Augustyn
• [Asterisk-Users] MySQL to static .conf   Juan Luis Moyano
• [Asterisk-Users] Polycom 500 Sound Problem   cyberguy at voicenet.com
• [Asterisk-Users] Re: app_valetparking.c for * STABLE (1.0.X)   Paul Zimm
• [Asterisk-Users] QuadBRI: How to set the outgoing callerid (KPN - NL)   Stijn Jonker
• [Asterisk-Users] PBXfreeware.org Open for business! / JavaScript module for Asterisk Unveiled!   Brian West
• [Asterisk-Users] Zaptel HEAD with * Stable?   Florian Overkamp
• [Asterisk-Users] RCAPI ISDN Support   dmatos at eddil.net
• [Asterisk-Users] outgoing call routing   Jose Vicente Ortega
• [Asterisk-Users] QuadBRI: How to set the outgoing callerid (KPN - NL)   Remco Barende
• [Asterisk-Users] SIP Ad-Hoc Conferencing with Asterisk   Joerg Furrer
• [Asterisk-Users] QuadBRI: How to set the outgoing callerid (KPN - NL)   Stijn Jonker
• [Asterisk-Users] Suggestions for using AbsoluteTimeout   Tim Connolly
• [Asterisk-Users] QuadBRI: How to set the outgoing callerid (KPN - NL)   Michiel van Baak
• [Asterisk-Users] RCAPI ISDN Support   Armin Schindler
• [Asterisk-Users] Re: Debian Vs Fedora   Chris Earle
• [Asterisk-Users] tos problem   Calin Serbanescu
• [Asterisk-Users] QuadBRI: How to set the outgoing callerid (KPN - NL)   Stijn Jonker
• [Asterisk-Users] Automatic Agent Login   Hendrik Magilsen
• [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.   qrss
• [Asterisk-Users] Re: app_valetparking.c for * STABLE   Nenad Radosavljevic
• [Asterisk-Users] Looking for PRI Outbound Caller ID Configuration   Shaun Tierney
• [Asterisk-Users] Polycom 500 Sound Problem   cyberguy at voicenet.com
• [Asterisk-Users] QuadBRI: How to set the outgoing callerid (KPN - NL)   Michiel van Baak
• [Asterisk-Users] outgoing call routing   Sergio Serrano
• [Asterisk-Users] SIP Ad-Hoc Conferencing with Asterisk   Pavel Jezek
• [Asterisk-Users] QuadBRI: How to set the outgoing callerid (KPN - NL)   Remco Barende
• [Asterisk-Users] AGI/PHP errors   Moises Silva
• [Asterisk-Users] How can you check that eg TDM04B hardware installed and drivers OK   Angus Comber
• [Asterisk-Users] VoiceMail   Carlos Rojas
• [Asterisk-Users] tos problem   Rich Adamson
• [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.   qrss
• [Asterisk-Users] PBXfreeware.org Open for business! / JavaScript module for Asterisk Unveiled!   David John Walsh
• [Asterisk-Users] Compilation Problem with asterisk-addons   Nico Giefing
• [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.   Rich Adamson
• [Asterisk-Users] Asterisk does not function without a DNS server   Eric Bishop
• [Asterisk-Users] QuadBRI: How to set the outgoing callerid (KPN - NL)   Stijn Jonker
• [Asterisk-Users] Contexts Calling Each Other   info at beprojects.com
• [Asterisk-Users] ee1000 Ethernet in Dell 1850   Rod Bacon
• [Asterisk-Users] Looking for PRI Outbound Caller ID Configuration   Shaun Tierney
• [Asterisk-Users] Looking for PRI Outbound Caller ID Configuration   Rich Adamson
• [Asterisk-Users] voicemail system   harry gaillac
• [Asterisk-Users] Panasonic KX-TD1232   Dan Morin
• [Asterisk-Users] Zaptel Disconnect Tone   Dan Morin
• [Asterisk-Users] TxFax: can't get a fax to destination (log inside)   Steve Underwood
• [Asterisk-Users] service scan   Moises Silva
• [Asterisk-Users] Automatic Agent Login   Dan Morin
• [Asterisk-Users] service scan   Dan Morin
• [Asterisk-Users] Re: app_valetparking.c for * STABLE (1.0.X)   David John Walsh
• [Asterisk-Users] AgentCallBacklogin (logout continued...)   1 2
• [Asterisk-Users] service scan   info at beprojects.com
• [Asterisk-Users] Prepaid systems considerations   Ricardo Poppi
• [Asterisk-Users] Re: Libtiff 3.5.7 - recommended version for spandsp   Roger Schreiter
• [Asterisk-Users] service scan   Chris Stone
• [Asterisk-Users] Re: Libtiff 3.5.7 - recommended version for spandsp   Brian West
• [Asterisk-Users] IAXy G729 Translation   Samy Antoun
• [Asterisk-Users] Unable to reconfigure channel   Jairo Barahona
• [Asterisk-Users] the table tariffrate is empty in Areskicc !   zhu
• [Asterisk-Users] TxFax: can't get a fax to destination (log inside)   Lee Howard
• [Asterisk-Users] Extension Configuration Best Practice   Anton Krall
• [Asterisk-Users] ee1000 Ethernet in Dell 1850   Orlando Guitián
• [Asterisk-Users] Looking for PRI Outbound Caller ID Configuration   Orlando Guitián
• [Asterisk-Users] voicemail system   Orlando Guitián
• [Asterisk-Users] Compilation Problem with asterisk-addons   Juan Luis Moyano
• [Asterisk-Users] accountcode not present in cdr   Max Clark
• [Asterisk-Users] Re: app_valetparking.c for * STABLE   Nenad Radosavljevic
• [Asterisk-Users] storing CDR records in a MySQL database   Joseph
• [Asterisk-Users] Unable to make outbound calls   Balaji NJL
• [Asterisk-Users] ee1000 Ethernet in Dell 1850   Andres
• [Asterisk-Users] accountcode not present in cdr   Andres
• [Asterisk-Users] asterisk and radius?   maka
• [Asterisk-Users] ee1000 Ethernet in Dell 1850   Andres
• [Asterisk-Users] Re: $0-per-month (pay as you go) provider with T.38?   Adam Megacz
• [Asterisk-Users] Asterisk does not function without a DNS ser ver   Guido Hecken
• [Asterisk-Users] [sourceforge.net abuse] New JAVA application server for Asterisk - OrderlyCalls   Adam Megacz
• [Asterisk-Users] Compilation Problem with asterisk-addons   Nico Giefing
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 131   Nguyen Trung Tin
• [Asterisk-Users] Panasonic KX-TD1232   Peter Svensson
• [Asterisk-Users] Re: Libtiff 3.5.7 - recommended version for spandsp   Marco Parmeggiani
• [Asterisk-Users] webvmail debian package   Tzafrir Cohen
• [Asterisk-Users] Debian Vs Fedora   Tzafrir Cohen
• [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation   David Wilson
• [Asterisk-Users] ee1000 Ethernet in Dell 1850   Zoa
• [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation   altus
• [Asterisk-Users] help for making several calls at the same time..   Erdem HAKİ
• [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation   David Wilson
• 回复： [Asterisk-Users] storing CDR records in a MySQL database   Gary Li
• [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation   altus
• [Asterisk-Users] Asterisk and Sirrix PCI4S0 echo cancellation   David Wilson
• [Asterisk-Users] ipswitchboard   altus
• [Asterisk-Users] OH323 with g723   Bashir Ullah - www.Lamsre.Com
• [Asterisk-Users] chan_h323 vs chan_oh323 & chan_ooh323   Bashir Ullah - www.Lamsre.Com
• [Asterisk-Users] voicemail system   harry gaillac
• [Asterisk-Users] Asterisk does not function without a DNS server   dsr at tao.merseine.nu
• [Asterisk-Users] Asterisk does not function without a DNS ser ver   Eric Bishop
• [Asterisk-Users] Problems with wew FXO modules for TDM400P   robert.brown01 at blueyonder.co.uk
• [Asterisk-Users] PLEASE HELP X100P no responding   Christian Callejon
• [Asterisk-Users] Looking for PRI Outbound Caller ID Configuration   Rich Adamson
• [Asterisk-Users] ee1000 Ethernet in Dell 1850   John Cianfarani
• [Asterisk-Users] Unable to reconfigure channel   Rich Adamson
• [Asterisk-Users] ee1000 Ethernet in Dell 1850   Daryl G. Jurbala
• [Asterisk-Users] Ground Start on Asterisk   Syed Akbar
• [Asterisk-Users] Looking for PRI Outbound Caller ID Configura tion   Huddleston, Robert
• [Asterisk-Users] Call file calling twice   Remco Barende
• [Asterisk-Users] Extension Configuration Best Practice   Rich Adamson
• [Asterisk-Users] Transfer   Victor Alvarez
• [Asterisk-Users] ee1000 Ethernet in Dell 1850   Dave Cotton
• [Asterisk-Users] Re: $0-per-month (pay as you go) provider with T.38?   Joshua Colp
• [Asterisk-Users] Transfer   Nabeel Jafferali
• SV: SV: [Asterisk-Users] Presence and IM?   Bjørn Ove Kristiansen
• [Asterisk-Users] Asterisk does not function without a DNS ser ver   Rich Adamson
• [Asterisk-Users] ast_data help   harry gaillac
• [Asterisk-Users] Ground Start on Asterisk   Doug Lytle
• [Asterisk-Users] Transfer   sylvain garcia
• [Asterisk-Users] Transfer   Jan Saell
• SV: SV: [Asterisk-Users] Presence and IM?   Bjørn Ove Kristiansen
• [Asterisk-Users] How can you check that eg TDM04B hardware installed and drivers OK   Rich Adamson
• [Asterisk-Users] MFC R2 - Can this be solved???   j_amorim
• SV: SV: [Asterisk-Users] Presence and IM?   Joshua Colp
• [Asterisk-Users] ast_data help   Joshua Colp
• 回复： [Asterisk-Users] storing CDR records in a MySQL database   Joseph
• [Asterisk-Users] Extension Configuration Best Practice   Joshua Colp
• [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.   Rich Adamson
• [Asterisk-Users] MFC R2 - Can this problem be solved??????????   j_amorim
• [Asterisk-Users] Extension Configuration Best Practice   Anton Krall
• [Asterisk-Users] modprobe wctdm waiting for ever   Edgardo Lust
• [Asterisk-Users] Cisco 7750   Mark Johnson
• SV: SV: [Asterisk-Users] Presence and IM?   Anton Krall
• [Asterisk-Users] Extension Configuration Best Practice   Anton Krall
• [Asterisk-Users] Automatic Agent Login   Waldo Rubinstein
• [Asterisk-Users] app_changrab.c released on pbxfreeware.org   Brian West
• [Asterisk-Users] How can you check that eg TDM04B hardware installed and drivers OK   Mike M
• [Asterisk-Users] Extension Configuration Best Practice   Joshua Colp
• SV: SV: SV: [Asterisk-Users] Presence and IM?   Bjørn Ove Kristiansen
• SV: SV: [Asterisk-Users] Presence and IM?   Bjørn Ove Kristiansen
• [Asterisk-Users] chan_unicall, bug in 1.0.X - 99% CPU   Gerardo Perosio
• [Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?   Robert Rozman
• [Asterisk-Users] MFC R2 - Can this problem be solved??????????   Steve Underwood
• [Asterisk-Users] Extension Configuration Best Practice   Goolsby, Daniel S (Daniel)
• [Asterisk-Users] ee1000 Ethernet in Dell 1850   Rich Adamson
• [Asterisk-Users] Ground Start on Asterisk   Rich Adamson
• SV: SV: [Asterisk-Users] Presence and IM?   Joshua Colp
• [Asterisk-Users] ast_data help   Matthew Boehm
• [Asterisk-Users] app_changrab.c released on pbxfreeware.org   Anton Krall
• [Asterisk-Users] Extension Configuration Best Practice   Anton Krall
• [Asterisk-Users] Extension Configuration Best Practice   Rich Adamson
• [Asterisk-Users] Extension Configuration Best Practice   Anton Krall
• [Asterisk-Users] Cisco 7750   Trey Scarborough
• [Asterisk-Users] Ground Start on Asterisk   Doug Lytle
• [Asterisk-Users] modprobe wctdm waiting for ever   Mike M
• [Asterisk-Users] Asterisk died - exactly every 60 minutes   Ronald Wiplinger
• [Asterisk-Users] Re: SNOM, Asterisk and Attended transfer (bug?)   Steve Davies
• [Asterisk-Users] Extension Configuration Best Practice   Rich Adamson
• [Asterisk-Users] ast_data help   harry gaillac
• [Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?   Steve Underwood
• [Asterisk-Users] asterisk-api   gale81 at virgilio.it
• [Asterisk-Users] Cisco 7750   Mark Johnson
• [Asterisk-Users] communication between IAX softphones   Marco Parmeggiani
• [Asterisk-Users] Extension Configuration Best Practice   steve szmidt
• [Asterisk-Users] app_changrab.c released on pbxfreeware.org   Dave Cotton
• [Asterisk-Users] Best Echo Canceller.   Chris Modesitt
• [Asterisk-Users] MeetMe Problems   Waldo Rubinstein
• [Asterisk-Users] Asterisk died - exactly every 60 minutes   Sebastian Silva
• SV: SV: [Asterisk-Users] Presence and IM?   Anton Krall
• [Asterisk-Users] [ot] wifi 3G gsm phone   trixter http://www.0xdecafbad.com
• [Asterisk-Users] app_changrab.c released on pbxfreeware.org   Anton Krall
• [Asterisk-Users] Extension Configuration Best Practice   Jay Milk
• [Asterisk-Users] accountcode not present in cdr   Max Clark
• [Asterisk-Users] accountcode not present in cdr   Max Clark
• [Asterisk-Users] ee1000 Ethernet in Dell 1850   Adam Robins
• [Asterisk-Users] Re: New JAVA application server for Asterisk - OrderlyCalls   Matt King
• SV: SV: [Asterisk-Users] Presence and IM?   Michiel van Baak
• [Asterisk-Users] Extension Configuration Best Practice   Joshua Colp
• [Asterisk-Users] Multiple Sipura 3000   Mike Benoit
• [Asterisk-Users] ast_data help   harry gaillac
• [Asterisk-Users] ee1000 Ethernet in Dell 1850   Orlando Guitián
• [Asterisk-Users] Best Echo Canceller.   Kris Boutilier
• [Asterisk-Users] Extension Configuration Best Practice   Denis Galvão - iSolve
• [Asterisk-Users] Extension Configuration Best Practice   Anton Krall
• [Asterisk-Users] Polycom and CallerID   Johann
• [Asterisk-Users] Asterisk answers with high pitch sound   CityTechs.Net
• [Asterisk-Users] chan_unicall, bug in 1.0.X - 99% CPU   Gerardo Perosio
• [Asterisk-Users] Grandstream 100 pricing question   Francesco Peeters
• [Asterisk-Users] QuadBRI: How to set the outgoing callerid (KPN - NL)   Stijn Jonker
• [Asterisk-Users] Asterisk in India?   Matthew Gibson
• [Asterisk-Users] Intermittent audio issues with Asterisk behind symmetrical firewa ll   Geoff Manning
• [Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?   Leandro Morgado
• [Asterisk-Users] Problem with Connecting PBX to Asterisk   Karthik Natarajan
• [Asterisk-Users] Astricon Europe Media Post!   Kristian Kielhofner
• [Asterisk-Users] 403 forbidden on SIP register   snacktime
• [Asterisk-Users] QuadBRI: How to set the outgoing callerid (KPN - NL)   Michiel van Baak
• [Asterisk-Users] MFC R2 - Can this problem be solved??????????   j_amorim
• [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.   qrss
• [Asterisk-Users] Digium Card: Echo, Echo and more Echo   Matthew Boehm
• [Asterisk-Users] Extension Configuration Best Practice   Andrew Kohlsmith
• [Asterisk-Users] Problem with Connecting pbx and asterisk: using TE405P Asterisk -> T1 -> PBX   Karthik Natarajan
• [Asterisk-Users] Ground Start on Asterisk   Rich Adamson
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 136   Nguyen Trung Tin
• [Asterisk-Users] Extension Configuration Best Practice   Robert Goodyear
• [Asterisk-Users] Asterisk answers with high pitch sound   Huddleston, Robert
• [Asterisk-Users] app_changrab.c released on pbxfreeware.org   Nenad Radosavljevic
• [Asterisk-Users] MeetMe Problems   Moises Silva
• [Asterisk-Users] Re: New JAVA application server for Asterisk - OrderlyCalls   Adam Megacz
• [Asterisk-Users] ee1000 Ethernet in Dell 1850   Andres
• [Asterisk-Users] Grandstream 100 pricing question   Kristof Hardy
• [Asterisk-Users] Problem with Connecting PBX to Asterisk   qrss
• [Asterisk-Users] NVFaxdetect   Eric Rees
• [Asterisk-Users] Extension Configuration Best Practice   Rich Adamson
• [Asterisk-Users] Extension Configuration Best Practice   Rich Adamson
• [Asterisk-Users] 403 forbidden on SIP register   Michiel van Baak
• [Asterisk-Users] SpanDSP+astfax with multiple fax pages   Daniel Cubero Salas, Ing.
• [Asterisk-Users] Digium Card: Echo, Echo and more Echo   Kris Boutilier
• [Asterisk-Users] Asterisk answers with high pitch sound   info at beprojects.com
• [Asterisk-Users] Re: New JAVA application server for Asterisk - OrderlyCalls   Matt King
• [Asterisk-Users] voip-info.org unreliable lately?   Damon Estep
• [Asterisk-Users] Asterisk died - exactly every 60 minutes   Ronald Wiplinger
• [Asterisk-Users] PBXfreeware.org Open for business! / JavaScript module for Asterisk Unveiled!   Brian West
• [Asterisk-Users] NVFaxdetect   Joseph
• [Asterisk-Users] chan_unicall and /dev/zap/channel   Gerardo Perosio
• [Asterisk-Users] IAX protocol will not go through firewall after certain time.   Joseph
• [Asterisk-Users] Zombie?   Chris Mason (Lists)
• [Asterisk-Users] logged in agent make an outbound call?   Damon Estep
• [Asterisk-Users] H323 on ovislink   cristian at redux.com.mx
• [Asterisk-Users] 403 forbidden on SIP register   snacktime
• [Asterisk-Users] NVFaxdetect   Jason Becker
• [Asterisk-Users] 403 forbidden on SIP register   Rich Adamson
• [Asterisk-Users] Polycom and CallerID   Chris Coulthurst
• [Asterisk-Users] FXS   Alessandro
• [Asterisk-Users] NVFaxdetect   Eric Rees
• [Asterisk-Users] Re: New JAVA application server for Asterisk - OrderlyCalls   Andrew Kohlsmith
• [Asterisk-Users] 403 forbidden on SIP register   snacktime
• [Asterisk-Users] Asterisk does not function without a DNS ser ver   Eric Bishop
• [Asterisk-Users] Digium Card: Echo, Echo and more Echo   Greg Boehnlein
• [Asterisk-Users] FXS   Mike M
• [Asterisk-Users] Echo Issues   Callum McGillivray
• [Asterisk-Users] Questions about FXO Outgoing Dialing   Matt Basch
• [Asterisk-Users] Seeking Inbound 800# Origination for Unique Prostate Cancer Support Call-In Show   Lee Barken
• [Asterisk-Users] PBXFreeware.org new res_js example order status checking script.   Brian West
• [Asterisk-Users] Echo Issues   David Phelan
• [Asterisk-Users] gxp-2000 tftp cfg   The VoIP Connection
• [Asterisk-Users] voip-info.org unreliable lately?   Brian West
• [Asterisk-Users] FXS   a.l.e
• [Asterisk-Users] Help on installing h323   craz sead
• [Asterisk-Users] Echo Issues   Callum McGillivray
• [Asterisk-Users] voip-info.org unreliable lately?   Eric Rees
• [Asterisk-Users] Echo Issues   Kris Boutilier
• [Asterisk-Users] logged in agent make an outbound call?   Asterisk
• [Asterisk-Users] gxp-2000 tftp cfg   Kristof Hardy
• [Asterisk-Users] voip-info.org unreliable lately?   Kristof Hardy
• [Asterisk-Users] tdm400p not working after cvs-head update   Paradise Dove
• [Asterisk-Users] Grandstream 100 pricing question   Francesco Peeters
• [Asterisk-Users] DID not working? + sendmail problems   Rick
• [Asterisk-Users] ASTERISK+SER+MWI   harry gaillac
• [Asterisk-Users] Help on installing h323   Nardis Dome
• [Asterisk-Users] 3-way conference using zap channels -- how is it done?   Dimitris Kouimintzis
• [Asterisk-Users] zeroconf help   stevanus
• [Asterisk-Users] Core Dump   Chee Foong
• [Asterisk-Users] Grandstream 100 pricing question   Ming-Wei Shih
• [Asterisk-Users] 3month Internship between February end July 2006   thomas DEILLON
• [Asterisk-Users] Grandstream 100 pricing question   Francesco Peeters
• [Asterisk-Users] using DBGet inside extensions.ael   Chris Stenton
• [Asterisk-Users] Sip client   gale81 at virgilio.it
• [Asterisk-Users] PPPD problem please help   Daniel Nyström
• [Asterisk-Users] call divert to TRUNK , if one number is unregistered?   Erdem HAKİ
• [Asterisk-Users] Asterisk to NEC NEAX   Ramkumar
• [Asterisk-Users] Group/Broadcast Voicemail   Zen Kato
• [Asterisk-Users] Sip client   Dave Walker
• [Asterisk-Users] Grandstream 100 pricing question   Pavel Jezek
• [Asterisk-Users] Spanish doc   Leonardo F. Bauchwitz
• [Asterisk-Users] Variables to emailbody of voicemail   Gabriel Perez S.
• [Asterisk-Users] ZapRAS   Daniel Nyström
• [Asterisk-Users] Dialplan Q: Dialing with Capi   Patrik Schindler
• [Asterisk-Users] Detecting the active queue agent...   Guilherme Lopes
• [Asterisk-Users] OT: Asterisk and Mambo - help wanted   Kristian Kielhofner
• [Asterisk-Users] Dialplan Q: Dialing with Capi   jurczak
• [Asterisk-Users] Re: app_changrab.c released on pbxfreeware.org   Nenad Radosavljevic
• [Asterisk-Users] Grandstream 100 pricing question   Peter Svensson
• [Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?   Peter Svensson
• [Asterisk-Users] is sip:%2321 valid invite?   Domjan Attila
• [Asterisk-Users] call divert to TRUNK ,if one number is unregistered?   Mohamed A. Gombolaty
• [Asterisk-Users] Is this server sufficient?   Francesco Peeters
• [Asterisk-Users] Is this server sufficient?   Dean Collins
• [Asterisk-Users] Is this server sufficient?   Francesco Peeters
• [Asterisk-Users] Is this server sufficient?   Andrew Latham
• [Asterisk-Users] meetme problem   Felix Skwarczynski
• [Asterisk-Users] Is this server sufficient?   Simone Cittadini
• [Asterisk-Users] Is this server sufficient?   Lists
• [Asterisk-Users] How can you check that eg TDM04B hardwareinstalled and drivers OK   Angus Comber
• [Asterisk-Users] meetme mute status   bdz
• [Asterisk-Users] Re: [Serusers] ASTERISK+SER+MWI   harry gaillac
• [Asterisk-Users] DID not working? + sendmail problems   Pascal Mosimann
• [Asterisk-Users] Re: [Serusers] ASTERISK+SER+MWI   Iqbal
• [Asterisk-Users] How can you check that eg TDM04B hardwareinstalled and drivers OK   John Novack
• [Asterisk-Users] Re: meetme mute status   Tony Mountifield
• [Asterisk-Users] TDM400P and Dell Poweredge 1750   Adam Robins
• [Asterisk-Users] How can you check that eg TDM04B hardwareinstalled and drivers OK   Angus Comber
• [Asterisk-Users] Dialplan Q: Dialing with Capi   Patrik Schindler
• [Asterisk-Users] gsm gateway   David Hajek
• [Asterisk-Users] Asterisk ended with exit status 139   Ronald Wiplinger
• [Asterisk-Users] Re: New JAVA application server for Asterisk - OrderlyCalls   Tom Rymes
• [Asterisk-Users] FXS interfaces   Alessandro
• [Asterisk-Users] call divert to TRUNK , if one number is unregistered?   Erdem HAKİ
• [Asterisk-Users] Grandstream 100 pricing question   barton-lists at gcmcomputers.com
• [Asterisk-Users] DID not working? + sendmail problems   Syed Akbar
• [Asterisk-Users] Polycom and CallerID   Johann
• [Asterisk-Users] Telrad + E&M T1 Trunk   Karthik Natarajan
• [Asterisk-Users] FOP related questions   Daniel ANDRE
• [Asterisk-Users] Problem with Connecting PBX to Asterisk   Karthik Natarajan
• [Asterisk-Users] 403 forbidden on SIP register   Rich Adamson
• [Asterisk-Users] logged in agent make an outbound call?   Damon Estep
• [Asterisk-Users] meetme mute status   Moises Silva
• [Asterisk-Users] Problem with Connecting PBX to Asterisk   qrss
• [Asterisk-Users] Spanish doc   Moises Silva
• [Asterisk-Users] voip-info.org unreliable lately?   Damon Estep
• [Asterisk-Users] Fwd:protocol TCP/UDP question   lucape at inwind.it
• [Asterisk-Users] Re: New JAVA application server for Asterisk - OrderlyCalls   Adam Goryachev
• [Asterisk-Users] Extension Configuration Best Practice   Denis Galvão - iSolve
• [Asterisk-Users] logged in agent make an outbound call?   Asterisk
• [Asterisk-Users] MeetMe Problems   Waldo Rubinstein
• [Asterisk-Users] Asterisk Manager Api   gale81 at virgilio.it
• [Asterisk-Users] Asterisk Manager Api   gale81 at virgilio.it
• [Asterisk-Users] New Asterisk Implementation   Don Brearley
• [Asterisk-Users] TDM400P DevKit Problem   Hamdy Elbarkouky
• [Asterisk-Users] TDM400P DevKit Problem   Hamdy Elbarkouky
• [Asterisk-Users] (no subject)   lucape at inwind.it
• [Asterisk-Users] Fwd:protocol TCP/UDP question   Sergio Chersovani
• [Asterisk-Users] logged in agent make an outbound call?   Damon Estep
• [Asterisk-Users] Grandstream 100 pricing question   Francesco Peeters
• [Asterisk-Users] Grandstream 100 pricing question   Francesco Peeters
• SV: [Asterisk-Users] voip-info.org unreliable lately?   Bjørn Ove Kristiansen
• [Asterisk-Users] FXS interfaces   Mike M
• [Asterisk-Users] Asterisk Manager Api   Maxime Renaud
• [Asterisk-Users] Asterisk to NEC NEAX   Allen Niven
• [Asterisk-Users] New Asterisk Implementation   harry gaillac
• [Asterisk-Users] New Asterisk Implementation   Mike M
• [Asterisk-Users] How can you check that egTDM04B hardwareinstalled and drivers OK   Angus Comber
• [Asterisk-Users] New Asterisk Implementation   Alexander Lopez
• [Asterisk-Users] TDM400P & Channel Group   Adam Robins
• [Asterisk-Users] automated response   Steve Hearst
• [Asterisk-Users] Grandstream 100 pricing question   barton-lists at gcmcomputers.com
• [Asterisk-Users] TE110P Card   MSEYE
• [Asterisk-Users] volume "fading in and out"   Asterisk
• [Asterisk-Users] Garbled one-way audio only with ulaw   rsenykoff at harrislogic.com
• [Asterisk-Users] RE: Asterisk to NEC NEAX   Jason Kawakami
• [Asterisk-Users] ISDN (PRI) in the US and Redirect?   John Millican
• [Asterisk-Users] FXS interfaces   Alessandro
• [Asterisk-Users] Performance Monitoring.   someshwarak
• [Asterisk-Users] Grandstream 100 pricing question   Wai-Sun Chia
• [Asterisk-Users] OT: Asterisk and Mambo - help wanted   JD Austin
• [Asterisk-Users] problem compile   ferrocristos at tiscali.it
• [Asterisk-Users] New Asterisk Implementation   Matt Fredrickson
• [Asterisk-Users] RE: TDM400P & Channel Group   Jason Kawakami
• [Asterisk-Users] problem compile   Sebastian Silva
• [Asterisk-Users] DIAX 0.9.15a with GSM gateway functionality   Dan
• [Asterisk-Users] Can I dial a number from handset to pickup voicemail?   Angus Comber
• [Asterisk-Users] Asterisk Manager Api   Moises Silva
• [Asterisk-Users] logged in agent make an outbound call?   Asterisk
• [Asterisk-Users] Re: volume "fading in and out"   Tony Mountifield
• [Asterisk-Users] New Asterisk Implementation   Seamus Abshere
• [Asterisk-Users] New Asterisk Implementation   Andrew Latham
• [Asterisk-Users] New Asterisk Implementation   Andrew Latham
• [Asterisk-Users] New Asterisk Implementation   TC
• [Asterisk-Users] RE: TDM400P & Channel Group   Adam Robins
• [Asterisk-Users] Re: volume "fading in and out"   Asterisk
• [Asterisk-Users] Presentation Number   Antoine Courouble
• [Asterisk-Users] problem compile   Elio Rojano
• [Asterisk-Users] Help on installing h323   Leon Sun
• [Asterisk-Users] A Simple * Answering Machine w/ Caller Screening like the olden days   Richard Koch
• [Asterisk-Users] voip-info.org unreliable lately?   Jay Milk
• [Asterisk-Users] Seeking Inbound 800# Origination for Unique Prostate Cancer Support Call-In Show   Leon Sun
• [Asterisk-Users] Using HEAD version of Zaptel with Asterisk Stable Release   Syed Akbar
• [Asterisk-Users] FXS interfaces   Jerry
• [Asterisk-Users] Re: TDM400P DevKit Problem   Anatoliy Kounitskiy
• [Asterisk-Users] MeetMe Problems   Waldo Rubinstein
• [Asterisk-Users] Re: New JAVA application server for Asterisk - OrderlyCalls   Emanuele Pucciarelli
• [Asterisk-Users] FXS interfaces   Mike M
• [Asterisk-Users] ZapRAS   Jason McAffee
• [Asterisk-Users] Seeking Inbound 800# Origination for Unique Prostate Cancer Support Call-In Show   CM Rahman Jr.
• [Asterisk-Users] Group/Broadcast Voicemail   Robert Goodyear
• [Asterisk-Users] voip-info.org unreliable lately?   Walt Reed
• [Asterisk-Users] Can I dial a number from handset to pickup voicemail?   Time Bandit
• [Asterisk-Users] Weird ring back   David Wilson
• [Asterisk-Users] Zap POTS Line Problem calling outbound   Adam Robins
• [Asterisk-Users] FXS interfaces   Alessandro
• [Asterisk-Users] New Asterisk Implementation   Matt Fredrickson
• [Asterisk-Users] A Simple * Answering Machine w/ Caller Screening like the olden days   Jon Gabrielson
• [Asterisk-Users] combining calls from 2 queues   rkb at i.frys.com
• [Asterisk-Users] Wireless & Wireline Integration   Hugh L. Johnson
• [Asterisk-Users] combining calls from 2 queues   Seamus Abshere
• [Asterisk-Users] Spanish doc   Ing CIP Alejandro Celi Mariátegui
• [Asterisk-Users] voip-info.org unreliable lately?   Frank Mayhar
• [Asterisk-Users] Group/Broadcast Voicemail   Zen Kato
• [Asterisk-Users] Using HEAD version of Zaptel with Asterisk Stable Release   Rich Adamson
• [Asterisk-Users] Group/Broadcast Voicemail   Robert Goodyear
• [Asterisk-Users] combining calls from 2 queues   Asterisk
• [Asterisk-Users] Seeking Inbound 800# Origination for Unique Prostate Cancer Support Call-In Show   Lee Barken
• [Asterisk-Users] voip-info.org unreliable lately?   Walt Reed
• [Asterisk-Users] HooDaHek 0.1 Released   Nathan Pralle
• [Asterisk-Users] Seeking Inbound 800# Origination for Unique Prostate Cancer Support Call-In Show   Michael D Schelin
• [Asterisk-Users] Setup suggestions/ideas   jamesm
• [Asterisk-Users] Group/Broadcast Voicemail   Zen Kato
• [Asterisk-Users] Is anyone using VOIPREACH   Joel Jn-Francois
• [Asterisk-Users] FXS interfaces   Mike M
• [Asterisk-Users] Zap POTS Line Problem calling outbound   Rich Adamson
• [Asterisk-Users] FXS interfaces   Jerry
• [Asterisk-Users] Adit600-->Asterisk Via MGCP   Garrett Smith
• [Asterisk-Users] H323 on ovislink   cristian at redux.com.mx
• [Asterisk-Users] combining calls from 2 queues   Shaun Ewing
• [Asterisk-Users] Connecting extern telephones,   satchid
• [Asterisk-Users] Question on bridged calls   snacktime
• [Asterisk-Users] Seeking Inbound 800# Origination for UniqueProstate Cancer Support Call-In Show   Leon Sun
• [Asterisk-Users] Is anyone using VOIPREACH   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Asterisk Manager Interface Remote Buffer Overflow Vulnerability   trixter http://www.0xdecafbad.com
• [Asterisk-Users] indexing tables for dialing   Ipek Zivane
• [Asterisk-Users] asterisk authentication issue   Mike
• [Asterisk-Users] RE: res_cepstral   Memon, Nauman
• [Asterisk-Users] add-on mysql cmd   johnny chan
• [Asterisk-Users] combining calls from 2 queues   rkb at i.frys.com
• [Asterisk-Users] RE: res_cepstral   Brian Roy
• [Asterisk-Users] Garbled one-way audio only with ulaw   Chris Mason (Lists)
• [Asterisk-Users] Zap POTS Line Problem calling outbound   John Novack
• [Asterisk-Users] Zaptel + IBM OpenPower Servers   Ilan Rabinovitch
• [Asterisk-Users] DMS-500 CID NAME Problem   JR Richardson
• [Asterisk-Users] asterisk authentication issue   scott
• [Asterisk-Users] TDM400P & Channel Group   Adam Goryachev
• [Asterisk-Users] FXS interfaces   Adam Goryachev
• [Asterisk-Users] Sip Sidecar Options   Max Clark
• [Asterisk-Users] Garbled one-way audio only with ulaw   rsenykoff at harrislogic.com
• [Asterisk-Users] *77 does not work ..   Brian Watters
• [Asterisk-Users] asterisk authentication issue   scott
• [Asterisk-Users] Zaptel card AND Ztdummy together?   Rod Bacon
• [Asterisk-Users] Asterisk + Asterisk-Stat as a Employee Time Clock   Joseph
• [Asterisk-Users] Native Bridge   Anton Krall
• [Asterisk-Users] TDM400P & Channel Group   George Pajari
• [Asterisk-Users] Zaptel card AND Ztdummy together?   qrss
• [Asterisk-Users] Malformed/Missing.URL Error from CallManager   Charles Huang
• [Asterisk-Users] Error on installing oh323 on asterisk   Charles Huang
• [Asterisk-Users] *67 with Sipura 3000   Massimo De Nadal
• [Asterisk-Users] missing cdr records   Rosario Pingaro
• [Asterisk-Users] TDM400P & Channel Group   Adam Goryachev
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Massimo De Nadal
• [Asterisk-Users] Zaptel card AND Ztdummy together?   Rod Bacon
• [Asterisk-Users] OT: MAX TNT and PRI calling name (CNAM) facility message   Kevin Blackham
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Armin Schindler
• [Asterisk-Users] Is anyone using VOIPREACH   Luki
• [Asterisk-Users] ASTCC not making calls   Juan Luis Moyano
• [Asterisk-Users] indexing tables for dialing   Luki
• [Asterisk-Users] Loosing hair on connecting Panasonic PBX- * - Euroisdn Italy   Robert Rozman
• [Asterisk-Users] flash panel only works with IP address   Ohad.Levy at infineon.com
• [Asterisk-Users] ASTCC not making calls   Juan Luis Moyano
• [Asterisk-Users] Configuration Cisco FXO with asterisk   craz sead
• [Asterisk-Users] Routing calls by trunk?   Rick
• [Asterisk-Users] Error on installing oh323 on asterisk   craz sead
• [Asterisk-Users] SIP DID routing   snacktime
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Sergio Chersovani
• [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *- Euroisdn Italy   Justin Newman
• [Asterisk-Users] Loosing hair on connecting Panasonic PBX- * - Euroisdn Italy   Peter Svensson
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Elmar Haneke
• [Asterisk-Users] Re: ZapRAS   Daniel Nyström
• [Asterisk-Users] Cisco 7960 firmware upgrade promblems   Patrick Lidstone (Personal E-mail)
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Massimo De Nadal
• [Asterisk-Users] Welltech 4 Port FXO - Asterisk   Erik Espinoza
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Massimo De Nadal
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Armin Schindler
• [Asterisk-Users] avm c2 correct configuration for two p2p lines   Simone Cittadini
• [Asterisk-Users] MeetMe Problems   Waldo Rubinstein
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Massimo De Nadal
• [Asterisk-Users] Music on Hold Choppy   Mahmoud Badran
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Armin Schindler
• [Asterisk-Users] Server Load/Capacity   Waldo Rubinstein
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Klaus-Peter Junghanns
• [Asterisk-Users] Least Cost Routing   Daniel ANDRE
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Armin Schindler
• [Asterisk-Users] Server Load/Capacity   tim panton
• [Asterisk-Users] Music on Hold Choppy   jurczak
• [Asterisk-Users] Cisco 7960 firmware upgrade promblems   Geoff Manning
• [Asterisk-Users] Asterisk in India?   Alistair Cunningham
• [Asterisk-Users] Question on bridged calls   Rich Adamson
• [Asterisk-Users] MGCP Groups   Mark Johnson
• [Asterisk-Users] Asterisk with failover and load balancing   Mohamed A. Gombolaty
• [Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?   Steve Underwood
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Klaus-Peter Junghanns
• [Asterisk-Users] Management: Reload performace & Realtime performance ?   René Ott
• [Asterisk-Users] Zap lines-inbound,outbound calls intersect   Colin E. McDonald
• [Asterisk-Users] GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?   Steve Underwood
• [Asterisk-Users] TDM400P & Channel Group   Rich Adamson
• [Asterisk-Users] Zap lines-inbound,outbound calls intersect   Betül Gözlükoğlu
• [Asterisk-Users] Help with Dial multiple channels simultanously   Kib Eki
• [Asterisk-Users] FXS interfaces   Alessandro
• [Asterisk-Users] Legal Requirement for Digital PBX   Roland Welker
• [Asterisk-Users] Asterisk @ Home setup & Doc   Paul
• [Asterisk-Users] Legal Requirement for Digital PBX   Steve Underwood
• [Asterisk-Users] Help with Dial multiple channels simultanously   Asterisk
• [Asterisk-Users] FXS interfaces   Alessandro
• [Asterisk-Users] FXS interfaces   Rich Adamson
• [Asterisk-Users] Re: Segfault on restart   Alphonse Ogulla
• [Asterisk-Users] indexing tables for dialing   Jay Milk
• [Asterisk-Users] AGI to monitor conenction quality   Chris Mason (Lists)
• [Asterisk-Users] Re: combining calls from 2 queues   alan
• [Asterisk-Users] Legal Requirement for Digital PBX   Roland Welker
• [Asterisk-Users] SIP DID routing   denis at isolve.com.br
• [Asterisk-Users] BRI signalling Morocco   igil at europesip.com
• [Asterisk-Users] Polycom display variable   Kib Eki
• [Asterisk-Users] Help with Dial multiple channels simultanously   Kib Eki
• [Asterisk-Users] This cpu usage doesn't seem right.   Matthew Boehm
• [Asterisk-Users] dialtone conf.of Turkey for ata186 sip   Betül Gözlükoğlu
• [Asterisk-Users] combining calls from 2 queues   Johann
• RES: [Asterisk-Users] MFC R2 - Can this problem be solved??????????   j_amorim
• [Asterisk-Users] so many FXS ports :)   Seamus Abshere
• [Asterisk-Users] Always forward an extension?   C. Hatton Humphrey
• [Asterisk-Users] ChanSpy on Asterisk v1.0.7   Tim Karl
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Armin Schindler
• [Asterisk-Users] privacy manager   John Hill
• [Asterisk-Users] Polycom display variable   Kib Eki
• [Asterisk-Users] missing cdr records   Paul Traue, Jr.
• [Asterisk-Users] CallerID name lookup AGI script   Oswaldo Arratia
• [Asterisk-Users] New Asterisk Implementation   Don Brearley
• [Asterisk-Users] New Asterisk Implementation   Don Brearley
• [Asterisk-Users] Monitoring Sirrix quad BRI channels   David Wilson
• [Asterisk-Users] New Asterisk Implementation   Don Brearley
• [Asterisk-Users] MFC R2 - Can this problem be solved??????????   Matias G.
• [Asterisk-Users] Asterisk Manager Interface Remote Buffer Overflow Vulnerability   Brian West
• [Asterisk-Users] ChanSpy on Asterisk v1.0.7   Brian West
• [Asterisk-Users] Ant: Re: [Asterisk-biz] sipredirect question   Axel Schemberg
• [Asterisk-Users] ChanSpy on Asterisk v1.0.7   Lee Archer
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Klaus-Peter Junghanns
• [Asterisk-Users] OT: MAX TNT and PRI calling name (CNAM) facility message   Matt Fredrickson
• [Asterisk-Users] privacy manager   Brian West
• [Asterisk-Users] Connecting extern telephones,   Carlos Chavez
• [Asterisk-Users] Always forward an extension?   Brian West
• [Asterisk-Users] Asterisk Manager Interface Remote Buffer Overflow Vulnerability   Zoa
• [Asterisk-Users] flash panel only works with IP address   Carlos Chavez
• [Asterisk-Users] Cisco 7960 firmware upgrade promblems   Tarpo, Louie
• [Asterisk-Users] the table tariffrate is empty in Areskicc !   Areski K
• [Asterisk-Users] mini itx   jltaylor
• [Asterisk-Users] Server Load/Capacity   Waldo Rubinstein
• [Asterisk-Users] *77 does not work ..   Jorge Carrasquillo
• [Asterisk-Users] SER and Asterisk question   Iqbal
• [Asterisk-Users] SER and Asterisk question   Iqbal
• [Asterisk-Users] mini itx   Matt Gibson
• [Asterisk-Users] Re: Cisco 7960 firmware upgrade promblems   Patrick Lidstone (Personal E-mail)
• [Asterisk-Users] ASTCC not making calls   Juan Luis Moyano
• [Asterisk-Users] mini itx   Bob Goddard
• [Asterisk-Users] so many FXS ports :)   Andrew Latham
• [Asterisk-Users] Dial a specific priority   jamesm
• [Asterisk-Users] ASTCC not making calls   Darren Wiebe
• [Asterisk-Users] mini itx   Iain Young
• [Asterisk-Users] Question on bridged calls   snacktime
• [Asterisk-Users] mini itx   jltaylor
• [Asterisk-Users] CallerID name lookup AGI script   Jay Milk
• [Asterisk-Users] ASTCC not making calls   jltaylor
• [Asterisk-Users] Asterisk Manager Interface Remote BufferOverflow Vulnerability   Dean Collins
• [Asterisk-Users] Re: CLUELESS NEWBIE needs help making an outbound sip call to PSTN   Steve
• [Asterisk-Users] SER and Asterisk question   Steve Blair
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Armin Schindler
• [Asterisk-Users] ASTCC not making calls   Darren Wiebe
• [Asterisk-Users] so many FXS ports :)   Peter Svensson
• [Asterisk-Users] OT: Asterisk and Mambo - help wanted   Wilson Pickett
• [Asterisk-Users] PRI auto reset?   Jason Brown
• [Asterisk-Users] mini itx   Wiley Siler
• [Asterisk-Users] ASTCC not making calls   Juan Luis Moyano
• [Asterisk-Users] 12 FXO ports into Asterisk   Darren Wright
• [Asterisk-Users] missing cdr records   Moises Silva
• [Asterisk-Users] Using 2 x DSL   VoIP-PBX
• [Asterisk-Users] Management: Reload performace & Realtime performance ?   snacktime
• [Asterisk-Users] mini itx   Chris Mason (Lists)
• [Asterisk-Users] mini itx   BJ Weschke
• [Asterisk-Users] Argentina and Mexico DID's & Termination   Carlos A Maimone at GAUSS
• [Asterisk-Users] mini itx   Chris Mason (Lists)
• [Asterisk-Users] Question on bridged calls   Rich Adamson
• [Asterisk-Users] Pls Hlp - Sendmail handoff / relay to cable   Paul
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   denis at isolve.com.br
• [Asterisk-Users] Using 2 x DSL   Michael Welter
• [Asterisk-Users] Looking for Honduras DIDs, Origination, Termination   Justin Newman
• [Asterisk-Users] Asterisk 'losing' upstream provider registration state during small network outages.   Steve
• [Asterisk-Users] Chan_Woomera beta released at www.pbxfreeware.org   Brian West
• [Asterisk-Users] SpanDSP - Squished Faxes   Richard Cook
• [Asterisk-Users] Driving direction sent to callers mobile phone via text/sms ?   Mike Dent
• [Asterisk-Users] French Audio Files   Jean-Michel Hiver
• [Asterisk-Users] *77 does not work ..   Michael Stahl
• [Asterisk-Users] Using 2 x DSL   jltaylor
• [Asterisk-Users] missing cdr records   Rosario Pingaro
• [Asterisk-Users] Re: TDM400P DevKit Problem   Hamdy Elbarkouky
• [Asterisk-Users] Management: Reload performace & Realtimeperformance ?   Rene Ott
• [Asterisk-Users] Re: Cisco 7960 firmware upgrade promblems   Tom Rymes
• [Asterisk-Users] Argentina and Mexico DID's & Termination   Alex Vishnev
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Matt Fredrickson
• [Asterisk-Users] Asterisk @ Home setup & Doc   Tom Rymes
• [Asterisk-Users] 12 FXO ports into Asterisk   Rich Adamson
• [Asterisk-Users] call transfers   David Arendt
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   jltaylor
• [Asterisk-Users] Voicemail   MDM
• [Asterisk-Users] 12 FXO ports into Asterisk   Darren Wright
• [Asterisk-Users] FOP related questions   Seth Remington
• [Asterisk-Users] Asterisk 1.0.8 Released   Russell Bryant
• [Asterisk-Users] Driving direction sent to callers mobile phone viatext/sms ?   Dean Collins
• [Asterisk-Users] French Audio Files   Asterisk
• [Asterisk-Users] Asterisk 'losing' upstream provider registration state during small network outages.   Joseph
• [Asterisk-Users] Dialplan Question   Dan Morin
• [Asterisk-Users] Using 2 x DSL   Andrew Latham
• [Asterisk-Users] Driving direction sent to callers mobile phone viatext/sms ?   Mike Dent
• [Asterisk-Users] so many FXS ports :)   rkb at i.frys.com
• [Asterisk-Users] mini itx   Andrew Latham
• [Asterisk-Users] Management: Reload performace & Realtimeperformance ?   Brian West
• [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.   qrss
• [Asterisk-Users] Re: Cisco 7960 firmware upgrade promblems   Patrick Lidstone (Personal E-mail)
• [Asterisk-Users] 12 FXO ports into Asterisk   Rusty Shackleford
• [Asterisk-Users] mini itx   Wiley Siler
• [Asterisk-Users] Automatically setting mailbox on registration of SIP device by SIP device's line #   Jonathan Feally
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Brian West
• [Asterisk-Users] privacy manager   Michael J. Tubby G8TIC
• [Asterisk-Users] 12 FXO ports into Asterisk   Darren Wright
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Domjan Attila
• [Asterisk-Users] Re: Cisco 7960 firmware upgrade promblems   Tarpo, Louie
• [Asterisk-Users] 12 FXO ports into Asterisk   Jason Becker
• [Asterisk-Users] Asterisk server with remote monitoring capabilities   beonice
• [Asterisk-Users] 12 FXO ports into Asterisk   Jorge Mendoza
• [Asterisk-Users] Asterisk server with remote monitoring capabilities   William Boehlke
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Brian West
• [Asterisk-Users] Asterisk server with remote monitoring capab ilities   mattf
• [Asterisk-Users] Asterisk server with remote monitoring capabilities   Michael Welter
• [Asterisk-Users] Asterisk server with remote monitoring capabilities   Joseph
• [Asterisk-Users] 12 FXO ports into Asterisk   Doug Lytle
• [Asterisk-Users] Asterisk server with remote monitoring capabilities   beonice
• [Asterisk-Users] Using 2 x DSL   VoIP-PBX
• [Asterisk-Users] Asterisk server with remote monitoring capabilities   Joseph
• [Asterisk-Users] CallerID name lookup AGI script   Oswaldo Arratia
• [Asterisk-Users] Asterisk & Zoom x5v 5565   Nazareno Pereyra Lima
• [Asterisk-Users] CallerID name lookup AGI script   Oswaldo Arratia
• [Asterisk-Users] HDLC abort 6 error   dstensrud at worleyco.com
• [Asterisk-Users] H323 vs OH323 GK registration issues   Eduardo Kaftanski
• [Asterisk-Users] H323 vs OH323 GK registration issues   Leandro Tenorio
• [Asterisk-Users] voicemail   Michael Di Martino
• [Asterisk-Users] Re: Chan_Woomera beta released at www.pbxfreeware.org   Peter Nixon
• [Asterisk-Users] 12 FXO ports into Asterisk   Nabeel Jafferali
• [Asterisk-Users] voicemail   Seth Remington
• [Asterisk-Users] SPA841 Utterly Horrible, are there any good stun hardphones?   Gregory Wiktor - ADCom Corp.
• [Asterisk-Users] Voicemail recording cutoff when silent for 1 second   Michael Stahl
• [Asterisk-Users] *77 does not work ..   Brian Watters
• [Asterisk-Users] More IP address in bindaddr directive   Coufal Bohuslav
• [Asterisk-Users] Requirement for internal calls   Michi
• [Asterisk-Users] Requirement for internal calls   Rob Thomas
• [Asterisk-Users] realtime sip confusion   snacktime
• [Asterisk-Users] asterisk security issue   Ohad.Levy at infineon.com
• [Asterisk-Users] flash panel only works with IP address   Daniel ANDRE
• [Asterisk-Users] French Audio Files   Dave Cotton
• [Asterisk-Users] HDLC abort 6 error   Klaus-Peter Junghanns
• [Asterisk-Users] SpanDSP - Squished Faxes   Marco Parmeggiani
• [Asterisk-Users] French Audio Files   harry gaillac
• [Asterisk-Users] French Audio Files   Thierry Wehr
• [Asterisk-Users] French Audio Files   Jean-Michel Hiver
• [Asterisk-Users] SpanDSP - Squished Faxes   El Flynn
• [Asterisk-Users] French Audio Files   harry gaillac
• [Asterisk-Users] Re: TDM400P DevKit Problem   Anatoliy Kounitskiy
• [Asterisk-Users] Management: Reload performace &Realtimeperformance ?   Rene Ott
• [Asterisk-Users] chan_capi-cm-0.5.1 and CLIR   asterisk_on_oelf
• [Asterisk-Users] BRIstuff/QuadBRI problem: Ring requested on unconfigured channel 255/255 span 5   Bruno.Voigt at ic3s.de
• [Asterisk-Users] polycom soundpoint ip 300   harry gaillac
• [Asterisk-Users] BRIstuff/QuadBRI problem: Ring requested on unconfigured channel 255/255 span 5   Klaus-Peter Junghanns
• [Asterisk-Users] chan_capi-cm-0.5.1 and CLIR   Armin Schindler
• [Asterisk-Users] Whole configuration for SMS   Gireesh Hariharasubramani
• [Asterisk-Users] Voicemail   jurczak
• [Asterisk-Users] Voicemail   Peter Bowyer
• [Asterisk-Users] Using 2 x DSL   Remco Barende
• [Asterisk-Users] How to setup two Asterisk boxes - keeping theregistration   Haydn.Kemmery at lihir.com.pg
• [Asterisk-Users] Re: Cisco 7960 firmware upgrade promblems   Tom Rymes
• [Asterisk-Users] Dial peer preference   kurt x
• [Asterisk-Users] FOP related questions   Daniel ANDRE
• [Asterisk-Users] Errors on SuSE 9.3 default install.   Zoltan Szecsei
• [Asterisk-Users] French Audio Files   Dave Cotton
• [Asterisk-Users] FOP related questions   Tony Hoyle
• [Asterisk-Users] Errors on SuSE 9.3 default install.   Doug Lytle
• [Asterisk-Users] Asterisk & Zoom x5v 5565   Geoff Manning
• [Asterisk-Users] Dial peer preference   Damon Estep
• [Asterisk-Users] Asterisk server with remote monitoringcapabilities   Matt Schulte
• [Asterisk-Users] SendText   Christian Hiller
• [Asterisk-Users] realtime sip confusion   Matthew Boehm
• [Asterisk-Users] voicemail   Michael Di Martino
• [Asterisk-Users] Errors on SuSE 9.3 default install.   Zoltan Szecsei
• [Asterisk-Users] RTP session between two end users   Erdem HAKİ
• [Asterisk-Users] Asterisk server with remote monitoring capabilities   Rich Adamson
• [Asterisk-Users] wcte11xp hardlock problem   John Mylchreest
• [Asterisk-Users] French Audio Files   Jean-Michel Hiver
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Andrew Kohlsmith
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Andrew Kohlsmith
• [Asterisk-Users] Asterisk server with remote monitoringcapabilities   Chris Mason (Lists)
• [Asterisk-Users] voicemail   Robert Webb
• [Asterisk-Users] Asterisk 1.0.8 Released   Iqbal
• [Asterisk-Users] PRI auto reset?   Wilson Pickett
• [Asterisk-Users] Extensions Puzzle: Contexts Confligting with each other.   Joan Bautista
• [Asterisk-Users] RTP session between two end users   Steve Clark
• [Asterisk-Users] Whole configuration for SMS   Wilson Pickett
• [Asterisk-Users] French Audio Files   Time Bandit
• [Asterisk-Users] More IP address in bindaddr directive   Kevin P. Fleming
• [Asterisk-Users] SendText   Jerry Geis
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   denis at isolve.com.br
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Andrew Kohlsmith
• [Asterisk-Users] Management: Reload performace &Realtimeperformance ?   Brian West
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   denis at isolve.com.br
• [Asterisk-Users] More IP address in bindaddr directive   Coufal Bohuslav
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Andrew Kohlsmith
• [Asterisk-Users] Asterisk server with remote monitoring capabilities   Max W Blackmer Jr
• [Asterisk-Users] SendText   Christian Hiller
• [Asterisk-Users] Loosing hair on connecting Panasonic PBX- * - Euroisdn Italy   Emanuele Pucciarelli
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Matt Fredrickson
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Julio Arruda
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Julio Arruda
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Andrew Kohlsmith
• [Asterisk-Users] format_base64.c released on pbxfreeware.org   Brian West
• [Asterisk-Users] Exposing Zap Channels on Server A to be Used By Server B   Wiley Siler
• [Asterisk-Users] realtime sip confusion   snacktime
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Kevin P. Fleming
• [Asterisk-Users] Exposing Zap Channels on Server A to be Used By Server B   Colin Anderson
• [Asterisk-Users] flash panel only works with IP address   Carlos Chavez
• [Asterisk-Users] Distinctive Ring for Agents (Was: Re: Asterisk 1.0.8)   alan
• [Asterisk-Users] SpanDSP - Squished Faxes   Carlos Chavez
• [Asterisk-Users] SendText   Christian Hiller
• [Asterisk-Users] SpanDSP - Squished Faxes   Richard Cook
• [Asterisk-Users] Exposing Zap Channels on Server A to be Used By Server B   Robert Goodyear
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   denis at isolve.com.br
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   denis at isolve.com.br
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   denis at isolve.com.br
• [Asterisk-Users] SpanDSP - Squished Faxes   Brian West
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Robert Webb
• [Asterisk-Users] meetme mute status   bdz at mailshack.com
• [Asterisk-Users] Unable to open pseudo channel for timing... Sound may be choppy.   Joseph
• [Asterisk-Users] Asterisk 1.0.8 Released   harry gaillac
• [Asterisk-Users] MusicOnHold   Giordano Grandis
• [Asterisk-Users] SpanDSP - Squished Faxes   Barton Fisher
• [Asterisk-Users] SpanDSP - Squished Faxes   Andrew Kohlsmith
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Andrew Kohlsmith
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Michael D Schelin
• [Asterisk-Users] Set global variables without extension..   chouck
• [Asterisk-Users] Set global variables without extension..   chouck
• [Asterisk-Users] Asterisk server with remote monitoring capabilities   Mark Musone
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   denis at isolve.com.br
• [Asterisk-Users] Set global variables without extension..   Kevin P. Fleming
• [Asterisk-Users] New astGUIclient version released 1.1.4   mattf
• [Asterisk-Users] SpanDSP - Squished Faxes   Brian West
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   denis at isolve.com.br
• [Asterisk-Users] SpanDSP - Squished Faxes   George Pajari
• [Asterisk-Users] SendText   Michiel van Baak
• [Asterisk-Users] Help instaling PyAsterisk   KC
• [Asterisk-Users] Re: Cisco 7960 firmware upgrade promblems   Greg Oliver
• [Asterisk-Users] SpanDSP - Squished Faxes   Lee Howard
• [Asterisk-Users] Asterisk & Zoom x5v 5565   Nazareno Pereyra Lima
• [Asterisk-Users] app_valetparking.c   Terry Wilson
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Julio Arruda
• [Asterisk-Users] Playtones volume control?   John Lange
• [Asterisk-Users] Tellabs Echo Canceller   ewr at erols.com
• [Asterisk-Users] SpanDSP - Squished Faxes   Brian West
• [Asterisk-Users] Asterisk & Zoom x5v 5565   snacktime
• [Asterisk-Users] Exposing Zap Channels on Server A to be Used ByServer B   Wiley Siler
• [Asterisk-Users] UTStarCom F1000 SIP configuration   MF Hulber
• [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *- Euroisdn Italy   Robert Rozman
• [Asterisk-Users] Tellabs Echo Canceller   Kris Boutilier
• [Asterisk-Users] CallerID name lookup AGI script   J P Edmund
• [Asterisk-Users] Exposing Zap Channels on Server A to be Used ByServer B   Robert Goodyear
• [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *- Euroisdn Italy   Emanuele Pucciarelli
• [Asterisk-Users] Exposing Zap Channels on Server A to be UsedByServer B   Wiley Siler
• [Asterisk-Users] Qualify Frequency   Matthew Boehm
• [Asterisk-Users] Help installing PyAsterisk   KC
• [Asterisk-Users] SpanDSP - Squished Faxes   Rich Adamson
• [Asterisk-Users] app_valetparking.c   Nenad Radosavljevic
• [Asterisk-Users] Tellabs Echo Canceller   Michael D Schelin
• [Asterisk-Users] Qualify Frequency   Kris Boutilier
• [Asterisk-Users] SpanDSP - Squished Faxes   Kevin P. Fleming
• [Asterisk-Users] Tellabs Echo Canceller   Kris Boutilier
• [Asterisk-Users] SpanDSP+astfax with multiple fax pages   Joan Bautista
• [Asterisk-Users] H323 vs OH323 GK registration issues   Eduardo Kaftanski
• [Asterisk-Users] Tellabs Echo Canceller   Andrew Kohlsmith
• [Asterisk-Users] realtime sip confusion   snacktime
• [Asterisk-Users] H323 vs OH323 GK registration issues   Leandro Tenorio
• [Asterisk-Users] Monitoring Sirrix quad BRI channels   Nicolás Gudiño
• [Asterisk-Users] SpanDSP - Squished Faxes   Rich Adamson
• [Asterisk-Users] SpanDSP+astfax with multiple fax pages - PRINTING A TIFF FILE (brief guide)   Daniel Cubero Salas, Ing.
• [Asterisk-Users] Tellabs Echo Canceller   Michael D Schelin
• [Asterisk-Users] SpanDSP - Squished Faxes   Marcelo Pacheco
• [Asterisk-Users] UTStarcom F1000 WiFi IP Phone Review   weicheng jiang
• [Asterisk-Users] CallerID name lookup AGI script   Oswaldo Arratia
• [Asterisk-Users] Running 6 copies of Asterisk on my machine   Michael Stahl
• [Asterisk-Users] H323 with Asterisk   Sahil Gupta
• [Asterisk-Users] Asterisk with dual WAN router   Joseph
• [Asterisk-Users] Asterisk with dual WAN router   Luki
• [Asterisk-Users] so many FXS ports :)   Carlos
• [Asterisk-Users] so many FXS ports :)   Bryce Chidester
• [Asterisk-Users] Best platform   Dinesh Nair
• [Asterisk-Users] so many FXS ports :)   Carlos
• [Asterisk-Users] Asterisk 'losing' upstream provider registration state during small network outages.   Steve
• [Asterisk-Users] UTStarcom F1000 WiFi IP Phone Review   Steve
• [Asterisk-Users] FW: ZAP to SIP Dial Plan   Walid Azab
• [Asterisk-Users] FXS interfaces   Mike M
• [Asterisk-Users] FXS interfaces   Mike M
• [Asterisk-Users] OH323, RxFax and codecs   Filipe Murias
• [Asterisk-Users] isdn channels busy   Asterisk
• [Asterisk-Users] Asterisk & Zoom x5v 5565   Nazareno Pereyra Lima
• [Asterisk-Users] Asterisk 'losing' upstream provider registrationstate during small network outages.   jurczak
• [Asterisk-Users] Asterisk and Cisco CallManager Integration   Walid Azab
• [Asterisk-Users] * 1.0.8: no more reacting to callerid?   Daryl Jones
• [Asterisk-Users] Asterisk and Cisco CallManager Integration   info at beprojects.com
• [Asterisk-Users] Asterisk server with remote monitoring capabilities   Tzafrir Cohen
• [Asterisk-Users] callerid in forwarded call   E Fierro
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Tzafrir Cohen
• [Asterisk-Users] INBAND DTMF G729 ASTERISK   Joshua Colp
• [Asterisk-Users] Asterisk 'losing' upstream provider registrationstate during small network outages.   Steve
• [Asterisk-Users] Management: Reload performace & Realtime performance ?   Tzafrir Cohen
• [Asterisk-Users] Management: Reload performace & Realtime performance ?   Joshua Colp
• [Asterisk-Users] PLEASE HELP X100P no responding   Tzafrir Cohen
• [Asterisk-Users] help for odbc storage   harry gaillac
• [Asterisk-Users] Asterisk 'losing' upstream provider registration state during small network outages.   Brian West
• [Asterisk-Users] UTStarcom F1000 WiFi IP Phone Review   Denis Galvão - iSolve
• [Asterisk-Users] SIP registration fails with realtime   Erick Johnson
• [Asterisk-Users] callerid in forwarded call   Robert Goodyear
• [Asterisk-Users] Looking for link.exe to compile G729 codec   Obelix
• [Asterisk-Users] UTStarcom F1000 WiFi IP Phone Review   Dan Perik
• [Asterisk-Users] UTStarcom F1000 WiFi IP Phone Review   Denis Galvão - iSolve
• [Asterisk-Users] * 1.0.8: no more reacting to callerid?   Kevin P. Fleming
• [Asterisk-Users] Management: Reload performace & Realtimeperformance ?   Rene Ott
• [Asterisk-Users] * 1.0.8: no more reacting to callerid?   Kevin P. Fleming
• [Asterisk-Users] Management: Reload performace & Realtimeperformance ?   Joshua Colp
• [Asterisk-Users] * 1.0.8: no more reacting to callerid?   Kevin P. Fleming
• [Asterisk-Users] ASTCC not billing   Ade Agbero
• [Asterisk-Users] ASTCC not billing   Darren Wiebe
• [Asterisk-Users] iaxy over the public cloud   Michael Di Martino
• [Asterisk-Users] Asterisk and Cisco CallManager Integration   Greg Oliver
• [Asterisk-Users] RE: iaxy over the public cloud   Robert Webb
• [Asterisk-Users] UTStarcom F1000 WiFi IP Phone Review   Paul
• [Asterisk-Users] How to bridge 2 calls together   Andy Yap
• [Asterisk-Users] Asterisk and Cisco CallManager Integration   Shaun Ewing
• [Asterisk-Users] Looking for link.exe to compile G729 codec   Tony Hoyle
• [Asterisk-Users] iaxy device   Michael Di Martino
• [Asterisk-Users] Everyone is busy/congested at this time   Kumara Jayaweera
• [Asterisk-Users] * fax reliability between ISDN PRI and FXS ports   Andres Maduro
• [Asterisk-Users] Livevoip   Darren Wiebe
• [Asterisk-Users] Livevoip   Moody
• [Asterisk-Users] IPSwitchBoard version 0.120 released   Thorben Jensen
• [Asterisk-Users] ASTCC not billing   Juan Luis Moyano
• [Asterisk-Users] LiveVoip is Bankrupt   Andres
• [Asterisk-Users] LiveVoip is Bankrupt   Yair Hakak
• [Asterisk-Users] Livevoip   Luki
• [Asterisk-Users] LiveVoip is Bankrupt   Brian Capouch
• [Asterisk-Users] Looking for link.exe to compile G729 codec   Obelix
• [Asterisk-Users] Livevoip   Obelix
• [Asterisk-Users] handle wrong extensions in Dialplam   Mahmoud Badran
• [Asterisk-Users] * fax reliability between ISDN PRI and FXS ports   cguy at bigpond.net.au
• [Asterisk-Users] chan_capi-cm-0.5.1 and CLIR   asterisk_on_oelf
• [Asterisk-Users] LiveVoip is Bankrupt   Bashir Ullah - www.Lamsre.Com
• [Asterisk-Users] handle wrong extensions in Dialplam   C. Hatton Humphrey
• [Asterisk-Users] cdr and billing   Mahmoud Badran
• [Asterisk-Users] Asterisk RealTime Voicemail   harry gaillac
• [Asterisk-Users] cdr and billing   jurczak
• [Asterisk-Users] cdr and billing   Mahmoud Badran
• [Asterisk-Users] Bug in Mailman version 2.1.5   Khubeka JM
• [Asterisk-Users] cdr and billing   Emanuele Pucciarelli
• [Asterisk-Users] Livevoip   Rich Adamson
• [Asterisk-Users] * 1.0.8: no more reacting to callerid?   Michael J. Tubby G8TIC
• [Asterisk-Users] Building DUNDi?   Bret Wortman
• [Asterisk-Users] help regarding h323.conf   SYED ADEEL ALI
• [Asterisk-Users] LiveVoip is Bankrupt   Rich Adamson
• [Asterisk-Users] * fax reliability between ISDN PRI and FXS ports   Rich Adamson
• [Asterisk-Users] failure in writing in pattern (routes)   wassim darwish
• [Asterisk-Users] * 1.0.8: no more reacting to callerid?   Kevin P. Fleming
• [Asterisk-Users] Asterisk and Cisco CallManager Integration   Walid Azab
• [Asterisk-Users] LiveVoip is Bankrupt   Mark Musone
• [Asterisk-Users] Re: [Asterisk-Dev] chan_capi-cm-0.5 release announcement   Armin Schindler
• [Asterisk-Users] Livevoip   Darren Wiebe
• [Asterisk-Users] LiveVoip is Bankrupt   Darren Wiebe
• [Asterisk-Users] ASTCC not billing   Darren Wiebe
• [Asterisk-Users] Livevoip 800 Choppy Audio   John Kington
• [Asterisk-Users] Looking for link.exe to compile G729 codec   Tony Hoyle
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Stefan Gofferje
• [Asterisk-Users] zeroconf help   Matt Riddell
• [Asterisk-Users] 2 servers via PRI   Michael Di Martino
• [Asterisk-Users] Asterisk RealTime Voicemail   Matthew Boehm
• [Asterisk-Users] Asterisk Manager Interface Remote Buffer Overflow Vulnerability   Matt Riddell
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Khubeka JM
• [Asterisk-Users] Zaptel Disconnect Tone   Dan Morin
• [Asterisk-Users] CDR: source completed with sip domain   Rosario Pingaro
• [Asterisk-Users] DID in 513 Cincinnati   John Kington
• [Asterisk-Users] CDR: source completed with sip domain   jurczak
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Peter Bowyer
• [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *-Euroisdn Italy   Robert Rozman
• [Asterisk-Users] need help for configuring voicemail with db   harry gaillac
• [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *-Euroisdn Italy   Emanuele Pucciarelli
• [Asterisk-Users] teliax [Was: LiveVoip is Bankrupt]   Adam Megacz
• [Asterisk-Users] indexing tables for dialing   Ipek Zivane
• [Asterisk-Users] Horrible MeetMe performance   Andy Rosen
• [Asterisk-Users] LiveVoip is Bankrupt   Orlando Guitián
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Areski K
• [Asterisk-Users] TDM400P (TDM02B) ceased to work...   Francois Meehan
• [Asterisk-Users] LiveVoip is Bankrupt   Andrew Kohlsmith
• [Asterisk-Users] ASTCC not billing   Ade Agbero
• [Asterisk-Users] TDM400P (TDM02B) ceased to work...   Robert P. McKenzie
• [Asterisk-Users] help regarding h323.conf   SYED ADEEL ALI
• [Asterisk-Users] TDM400P (TDM02B) ceased to work...   Francois Meehan
• [Asterisk-Users] Horrible MeetMe performance   Dan Morin
• [Asterisk-Users] LiveVoip is Bankrupt   Orlando Guitián
• [Asterisk-Users] LiveVoip is Bankrupt   steve szmidt
• [Asterisk-Users] Prepaid for mysql and simple auth   Ricardo Poppi
• [Asterisk-Users] DID in 513 Cincinnati   Jeff Glassman
• [Asterisk-Users] LiveVoip is Bankrupt   trixter http://www.0xdecafbad.com
• [Asterisk-Users] LiveVoip is Bankrupt   trixter http://www.0xdecafbad.com
• [Asterisk-Users] handle wrong extensions in Dialplam   Wayne
• [Asterisk-Users] LiveVoip is Bankrupt   snacktime
• [Asterisk-Users] LiveVoip is Bankrupt   Alexander Lopez
• [Asterisk-Users] ASTCC not billing   Darren Wiebe
• [Asterisk-Users] TDM400P (TDM02B) ceased to work...   Francois Meehan
• [Asterisk-Users] Changing Caller ID   Jeff Glassman
• [Asterisk-Users] LiveVoip is Bankrupt   Matt Riddell
• [Asterisk-Users] LiveVoip is Bankrupt   Peter Corlett
• [Asterisk-Users] LiveVoip is Bankrupt   Joshua Colp
• [Asterisk-Users] Re: Horrible MeetMe performance   Tony Mountifield
• [Asterisk-Users] LiveVoip is Bankrupt   Andres
• [Asterisk-Users] Zaptel card AND Ztdummy together?   Rod Bacon
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   harry gaillac
• [Asterisk-Users] LiveVoip is Bankrupt   Listacc
• [Asterisk-Users] Changing Caller ID   Bryce Chidester
• [Asterisk-Users] LiveVoip is Bankrupt   Sahil Gupta
• [Asterisk-Users] LiveVoip is Bankrupt   trixter http://www.0xdecafbad.com
• [Asterisk-Users] LiveVoip is Bankrupt   trixter http://www.0xdecafbad.com
• [Asterisk-Users] LiveVoip is Bankrupt   snacktime
• [Asterisk-Users] Re: Horrible MeetMe performance   Dan Morin
• [Asterisk-Users] Re: Horrible MeetMe performance   qrss
• [Asterisk-Users] LiveVoip is Bankrupt   Rich Adamson
• [Asterisk-Users] Missing first second of voice on outgoing SIP/IAX calls   snacktime
• [Asterisk-Users] APP - ValetParking on CVS-HEAD -- instructions on its use, anyone?   Chris Coulthurst
• [Asterisk-Users] LiveVoip is Bankrupt   Marie
• [Asterisk-Users] Re: Horrible MeetMe performance   Rob Thomas
• [Asterisk-Users] Asterisk 'losing' upstream provider registration state during small network outages.   Marie
• [Asterisk-Users] LiveVoip is Bankrupt   Michael Di Martino
• [Asterisk-Users] TDM400P-04B fails after reboot   Andrew
• [Asterisk-Users] How can you check that egTDM04B hardwareinstalled and drivers OK   Rich Adamson
• [Asterisk-Users] HT-488 vs. SPA-3000?   John Cianfarani
• [Asterisk-Users] LiveVoip is Bankrupt   Rich Adamson
• [Asterisk-Users] LiveVoip is Bankrupt   trixter http://www.0xdecafbad.com
• [Asterisk-Users] How can you check that egTDM04B hardwareinstalled and drivers OK   Steve Totaro
• [Asterisk-Users] Re: Horrible MeetMe performance   Tzafrir Cohen
• [Asterisk-Users] Re: Horrible MeetMe performance   Brian Capouch
• [Asterisk-Users] FXO as modem (was: * fax reliability between ISDN PRI and FXS ports)   Jerry
• [Asterisk-Users] Asterisk and Cisco CallManager Integration   Vamsi Pottangi
• [Asterisk-Users] Re: Horrible MeetMe performance   Peter Svensson
• [Asterisk-Users] Monitoring Sirrix quad BRI channels   David Wilson
• [Asterisk-Users] newbie here ...... regarding h323.conf   Adeel -31
• [Asterisk-Users] Chan_Woomera beta released at www.pbxfreeware.org   Adam Goryachev
• [Asterisk-Users] No Sound at all   RockWater !
• [Asterisk-Users] LiveVoip is Bankrupt   Marcel van Kaam, Fonetica
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Francesco Peeters
• [Asterisk-Users] RTP session between two end users   Erdem HAKİ
• [Asterisk-Users] Re: Horrible MeetMe performance   Tony Mountifield
• [Asterisk-Users] Re: Horrible MeetMe performance   Tony Mountifield
• [Asterisk-Users] Re: Horrible MeetMe performance   Tony Mountifield
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Marcel van Kaam, Fonetica
• [Asterisk-Users] bristuff-0.2.0-RC8h does not compile   Klaus-Peter Junghanns
• [Asterisk-Users] isdn channels busy   Klaus-Peter Junghanns
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Emanuele Pucciarelli
• [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received   David Masure
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   harry gaillac
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Peter Bowyer
• [Asterisk-Users] MWI   Christian Hiller
• [Asterisk-Users] Strange behaviour with lost internet connection   Ola Lidholm
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   harry gaillac
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Armin Schindler
• [Asterisk-Users] Re: [Asterisk-Dev] chan_capi-cm-0.5 release announcement   Armin Schindler
• [Asterisk-Users] chan_capi-cm-0.5.2 fixup release   Armin Schindler
• [Asterisk-Users] Re: [Asterisk-Dev] chan_capi-cm-0.5 release announcement   Stefan Gofferje
• [Asterisk-Users] chan_capi-cm-0.5 release announcement   Stefan Gofferje
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   harry gaillac
• [Asterisk-Users] iaxy over the public cloud   Michael Di Martino
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Peter Bowyer
• [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received   Julian J. M.
• R: [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received   Giordano Grandis
• [Asterisk-Users] isdn channels busy   Asterisk
• [Asterisk-Users] Fw: linksys rt31p2 test case   Dionisis Koumouras
• [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received   Tzafrir Cohen
• [Asterisk-Users] FXO as modem (was: * fax reliability between ISDN PRI and FXS ports)   Andrew Kohlsmith
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Andrew Kohlsmith
• [Asterisk-Users] LiveVoip is Bankrupt   Andrew Kohlsmith
• [Asterisk-Users] Fw: linksys rt31p2 test case   jurczak
• [Asterisk-Users] LiveVoip is Bankrupt   Francesco Peeters
• [Asterisk-Users] Re: Horrible MeetMe performance   Rob Thomas
• [Asterisk-Users] Re: Horrible MeetMe performance   Rob Thomas
• [Asterisk-Users] Strange behaviour with lost internet connection   Rob Thomas
• [Asterisk-Users] Fw: linksys rt31p2 test case   Dionisis Koumouras
• [Asterisk-Users] facing troubles with routes patterns dialplan   wassim darwish
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   harry gaillac
• [Asterisk-Users] Failover Design   John Cianfarani
• [Asterisk-Users] Native MoH patch for 1.0.8?   Patrick
• [Asterisk-Users] Do includes include the includes   Arvanitis Kostas
• [Asterisk-Users] FXO as modem (was: * fax reliability between ISDN PRI and FXS ports)   Rich Adamson
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   harry gaillac
• [Asterisk-Users] realtime sip confusion   Steve Blair
• [Asterisk-Users] Howto answer, hold and transfer a incoming call?   Gunnar Henne
• [Asterisk-Users] TE100P   Sahil Gupta
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   John Goerzen
• [Asterisk-Users] FXO as modem (was: * fax reliability between ISDN PRI and FXS ports)   Rich Adamson
• [Asterisk-Users] TDM card and voicemail volume   Adam Robins
• [Asterisk-Users] Comedian Mail User Setup Prompts   Adam Robins
• [Asterisk-Users] Fw: linksys rt31p2 test case   Michael Stahl
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   Andrew Latham
• [Asterisk-Users] Comedian Mail User Setup Prompts   Asterisk
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   Dan Perik
• [Asterisk-Users] Accessing SIP username from AGI script   David Shirley
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Andrew Kohlsmith
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   Andrew Kohlsmith
• [Asterisk-Users] LiveVoip is Bankrupt   Andrew Kohlsmith
• [Asterisk-Users] Accessing SIP username from AGI script   jurczak
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   Andrew Latham
• R: [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received   Ivan Ricondo
• [Asterisk-Users] LiveVoip is Bankrupt   Rusty Shackleford
• [Asterisk-Users] IAXY setup   Michael Di Martino
• [Asterisk-Users] ???? WARNING[20313]: channel.c:531 ast_channel_walk_locked ????   niels at wxn.nl
• [Asterisk-Users] LogWatch for Asterisk   Michael Stahl
• [Asterisk-Users] LiveVoip is Bankrupt   Danny Froberg
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Peter Bowyer
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   asterisk at txpe.net
• [Asterisk-Users] Accessing SIP username from AGI script   Moises Silva
• [Asterisk-Users] Monitoring Sirrix quad BRI channels   David Wilson
• [Asterisk-Users] Asterisk RealTime Voicemail   Michael Stearne
• [Asterisk-Users] Re: FXO as modem (was: * fax reliability between ISDN PRI andFXS ports)   Jerry
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   Rick Baranowski
• [Asterisk-Users] LogWatch for Asterisk   Rick Baranowski
• [Asterisk-Users] OT: MAX TNT and PRI calling name (CNAM) facility message   Kevin Blackham
• [Asterisk-Users] LiveVoip is Bankrupt   Wiley Siler
• [Asterisk-Users] LiveVoip is Bankrupt   Wiley Siler
• [Asterisk-Users] Livevoip 800 Choppy Audio   Wiley Siler
• [Asterisk-Users] LiveVoip is Bankrupt   Wiley Siler
• [Asterisk-Users] tdm400p not working after cvs-head update   David Romero
• [Asterisk-Users] Asterisk and conference bridging...   carlos.rivas at sungard.com
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   asterisk at txpe.net
• [Asterisk-Users] LiveVoip is Bankrupt   Matt Riddell
• [Asterisk-Users] LiveVoip is Bankrupt   Wiley Siler
• [Asterisk-Users] LiveVoip is Bankrupt   Wiley Siler
• [Asterisk-Users] dropcount   David Hajek
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   harry gaillac
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   harry gaillac
• [Asterisk-Users] DID in Western Canada   Nelson Loyola
• [Asterisk-Users] realtime sip confusion   snacktime
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   John Goerzen
• [Asterisk-Users] LiveVoip is Bankrupt   Wiley Siler
• [Asterisk-Users] announced transfer   Markus Monka
• [Asterisk-Users] Passing called number in SIP   snacktime
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   Wiley Siler
• [Asterisk-Users] polycom soundpoint ip 300   Wilson Pickett
• [Asterisk-Users] LiveVoip is Bankrupt   Marcel van Kaam, Fonetica
• [Asterisk-Users] Bad Bad Performance; Max 20 Calls on Quad Proc?   Matthew Boehm
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   asterisk at txpe.net
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   Wiley Siler
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Andrew Kohlsmith
• [Asterisk-Users] RTP session between two end users   Eric Wieling aka ManxPower
• [Asterisk-Users] announced transfer   Flu
• [Asterisk-Users] LiveVoip is Bankrupt   Lists
• [Asterisk-Users] Chan_Woomera beta released at www.pbxfreeware.org   TC
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   Chad Osmond
• [Asterisk-Users] LogWatch for Asterisk   Moises Silva
• [Asterisk-Users] Bad Bad Performance; Max 20 Calls on Quad Proc?   Wiley Siler
• [Asterisk-Users] LiveVoip is Bankrupt   andrew matthews
• [Asterisk-Users] polycom soundpoint ip 300   Robert Goodyear
• [Asterisk-Users] LiveVoip is Bankrupt   David Brodbeck
• [Asterisk-Users] Comedian Mail User Setup Prompts   David Brodbeck
• [Asterisk-Users] TDM card and voicemail volume   David Brodbeck
• [Asterisk-Users] LiveVoip is Bankrupt   Michael Di Martino
• [Asterisk-Users] Asterisk and conference bridging...   Moises Silva
• [Asterisk-Users] polycom soundpoint ip 300   Wilson Pickett
• [Asterisk-Users] PSTN IAX Connections / Line Banks   Matthew H. Franz
• [Asterisk-Users] LiveVoip is Bankrupt   trixter http://www.0xdecafbad.com
• [Asterisk-Users] polycom soundpoint ip 300   harry gaillac
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   andrew matthews
• [Asterisk-Users] IAXY setup   Mojo with Horan & Company, LLC
• [Asterisk-Users] RE: [Serusers] *** SER - Asterisk   harry gaillac
• [Asterisk-Users] Re: Chan_Woomera beta released at www.pbxfreeware.org   Tony Mountifield
• [Asterisk-Users] Dialogic D/300pci-E1 card   Florin Mandache
• [Asterisk-Users] LiveVoip is Bankrupt   Andrew Kohlsmith
• [Asterisk-Users] H.323 (Asterisk@Home)   Dave Walker
• [Asterisk-Users] LiveVoip is Bankrupt   Andrew Kohlsmith
• [Asterisk-Users] LiveVoip is Bankrupt   Steven Kalcevich
• [Asterisk-Users] OT: Good soft-phone on Linux   Hamish Whittal
• [Asterisk-Users] LiveVoip is Bankrupt   Wiley Siler
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   steve szmidt
• [Asterisk-Users] RE: [Serusers] *** SER - Asterisk   Andrew Kohlsmith
• [Asterisk-Users] polycom soundpoint ip 300   harry gaillac
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   andrew matthews
• [Asterisk-Users] Dialogic D/300pci-E1 card   Eric Wieling aka ManxPower
• [Asterisk-Users] TDM card and voicemail volume   Andrew Kohlsmith
• [Asterisk-Users] LiveVoip is Bankrupt   trixter http://www.0xdecafbad.com
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   Mike
• [Asterisk-Users] AGI "say number" but in french   David John Walsh
• [Asterisk-Users] IAXY setup   Michael Di Martino
• [Asterisk-Users] Snom 220 Active Call Lights   Phil Smith
• [Asterisk-Users] LiveVoip is Bankrupt   Michael Di Martino
• [Asterisk-Users] LiveVoip is Bankrupt   Michael Di Martino
• [Asterisk-Users] OT: Good soft-phone on Linux   Wiley Siler
• [Asterisk-Users] Polycom & VPN trouble   Tim Pushor
• [Asterisk-Users] Dialogic D/300pci-E1 card   Francesco Peeters
• [Asterisk-Users] LiveVoip is Bankrupt   jurczak
• [Asterisk-Users] VoIP provider in Switzerland   Juan Jose Comellas
• [Asterisk-Users] PSTN IAX Connections / Line Banks   Huddleston, Robert
• [Asterisk-Users] LiveVoip is Bankrupt   Dan Perik
• [Asterisk-Users] Chan_Woomera beta released at www.pbxfreeware.org   Brian West
• [Asterisk-Users] Re: LiveVoip is Bankrupt   John Goerzen
• [Asterisk-Users] how to set agent to busy when agent makes a outgoing call?   Jon Gabrielson
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   harry gaillac
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   Michael Di Martino
• [Asterisk-Users] Dialogic D/300pci-E1 card   Francesco Peeters
• [Asterisk-Users] Dialogic D/300pci-E1 card   Florin Mandache
• [Asterisk-Users] RE: [Serusers] *** SER - Asterisk   Ben Blakely
• [Asterisk-Users] Re: Chan_Woomera beta released at www.pbxfreeware.org   Brian West
• [Asterisk-Users] LogWatch for Asterisk   snacktime
• [Asterisk-Users] Bad Bad Performance; Max 20 Calls on Quad Proc?   Asterisk
• [Asterisk-Users] LiveVoip is Bankrupt   Robert Webb
• [Asterisk-Users] Bridging and unbridging channels   Me
• [Asterisk-Users] ASTCC not billing   Darren Wiebe
• [Asterisk-Users] LiveVoip is Bankrupt   John Millican
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   Andrew Kohlsmith
• [Asterisk-Users] OT: Good soft-phone on Linux   Tzafrir Cohen
• [Asterisk-Users] LiveVoip is Bankrupt   Wiley Siler
• [Asterisk-Users] HooDaHek 0.2 Released   Nathan Pralle
• [Asterisk-Users] Just let whiners whine... Please?   Francesco Peeters
• [Asterisk-Users] Why this thread   Gary
• [Asterisk-Users] AGI "say number" but in french   Robert Goodyear
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Thierry Wehr
• [Asterisk-Users] LiveVoip is Bankrupt   Wiley Siler
• [Asterisk-Users] Possible bug in meetme when hangup   Jerry Geis
• [Asterisk-Users] Dialogic D/300pci-E1 card   Florin Mandache
• [Asterisk-Users] Chan_Woomera beta released at www.pbxfreeware.org   Tzafrir Cohen
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   Wiley Siler
• [Asterisk-Users] ASTCC not billing   Juan Luis Moyano
• [Asterisk-Users] Passing called number in SIP   Andres
• [Asterisk-Users] Level 3 SIP <--> asterisk   Jim Gottlieb
• [Asterisk-Users] LiveVoip is Bankrupt   trixter http://www.0xdecafbad.com
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   steve szmidt
• [Asterisk-Users] Passing called number in SIP   snacktime
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   Robert Webb
• [Asterisk-Users] HooDaHek 0.2 Released   Rusty Shackleford
• [Asterisk-Users] TDM card and voicemail volume   Steve Prior
• [Asterisk-Users] PSTN IAX Connections / Line Banks   John Novack
• [Asterisk-Users] Dialogic D/300pci-E1 card   Eric Wieling aka ManxPower
• [Asterisk-Users] SixTel?   JD Austin
• [Asterisk-Users] Chan_Woomera beta released at www.pbxfreeware.org   Brian West
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   Darren Wiebe
• [Asterisk-Users] LiveVoip is Bankrupt   Steve Totaro
• [Asterisk-Users] TE100P   Steve Totaro
• [Asterisk-Users] Asterisk server with remote monitoring capabilities   beonice
• [Asterisk-Users] How can you check that egTDM04Bhardwareinstalled and drivers OK   Steve Totaro
• [Asterisk-Users] is teliax down?   Adam Megacz
• [Asterisk-Users] Disable record busy greeting option in voicemail   Damon Estep
• [Asterisk-Users] DID in Western Canada   Leon Sun
• [Asterisk-Users] Livevoip 800 Choppy Audio   Rich Adamson
• [Asterisk-Users] Re: LiveVoip is Bankrupt   Steve Totaro
• [Asterisk-Users] Asterisk ended with exit status 1   Federico Alves
• [Asterisk-Users] Polycom & VPN trouble   gw at adcomcorp.com
• [Asterisk-Users] Native MoH patch for 1.0.8?   Juan Jose Comellas
• [Asterisk-Users] Polycom & VPN trouble   Tim Pushor
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   r00t
• [Asterisk-Users] Newbie Confusion on Call Forward and DBput/DBdel   Jeffrey Starin
• [Asterisk-Users] Eicon equipment, BRI Server or PRI?   gw at adcomcorp.com
• [Asterisk-Users] TDM card and voicemail volume   Rich Adamson
• [Asterisk-Users] LiveVoip is Bankrupt   Scott Miller
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   Rich Adamson
• [Asterisk-Users] Level 3 SIP <--> asterisk   Max Clark
• [Asterisk-Users] Asterisk server with remote monitoring capabilities   Max Clark
• [Asterisk-Users] How to get the outbound data of agent in queue   Gary Li
• [Asterisk-Users] Asterisk server with remote monitoring capabilities   Marie
• [Asterisk-Users] Fw: shoutcast mp3 music onhold with amp portal?   hank
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   Jay Milk
• [Asterisk-Users] is teliax down?   Jay Milk
• [Asterisk-Users] LiveVoip is Bankrupt   Jay Milk
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   harry gaillac
• [Asterisk-Users] LiveVoip is Bankrupt   Terry H. Gilsenan
• [Asterisk-Users] Can anyone guide me regarding h323.cong ???   Adeel -31
• [Asterisk-Users] H323   Ronald_Wiplinger
• [Asterisk-Users] SixTel?   Erik Espinoza
• [Asterisk-Users] Shoutcast Music On Hold problems?   hank
• [Asterisk-Users] Eicon equipment, BRI Server or PRI?   Armin Schindler
• [Asterisk-Users] polycom soundpoint ip 300   Wilson Pickett
• [Asterisk-Users] RTP session between two end users   Erdem HAKİ
• [Asterisk-Users] Passing called number in SIP   Andres
• [Asterisk-Users] H323   Tzafrir Cohen
• [Asterisk-Users] polycom soundpoint ip 300   harry gaillac
• [Asterisk-Users] Some phone numbers always busy   vdasilva
• [Asterisk-Users] cheap HFC card on Bristuff vs cheap HFC card on i4l vs Fritz ISDN BRI card on CAPI   vdasilva
• [Asterisk-Users] Re: H323   Tony Mountifield
• [Asterisk-Users] Passing called number in SIP   snacktime
• [Asterisk-Users] is teliax down?   Rick Baranowski
• [Asterisk-Users] H323   Ronald_Wiplinger
• [Asterisk-Users] Dialogic D/300pci-E1 card   tim panton
• [Asterisk-Users] Eicon equipment, BRI Server or PRI?   Clive
• [Asterisk-Users] HOW TO WRITE ROUTE PATTERNS DIALPLAN   wassim darwish
• [Asterisk-Users] AVM CAPI INSTALLATION   sylvain garcia
• [Asterisk-Users] Re: H323   Ronald_Wiplinger
• [Asterisk-Users] RE: [Serusers] *** SER - Asterisk   harry gaillac
• [Asterisk-Users] E100P configuration   Musaluke AK
• [Asterisk-Users] pbx_extension_helper: No application 'agi'   Tom Fielding
• [Asterisk-Users] cheap HFC card on Bristuff vs cheap HFC card oni4l vs Fritz ISDN BRI card on CAPI   David Masure
• [Asterisk-Users] simultaneus calls?   Erdem HAKİ
• [Asterisk-Users] Re: H323   Ronald_Wiplinger
• [Asterisk-Users] Re: H323   Damian Minkov
• [Asterisk-Users] GSM/PSTN Gateway function of DIAX - feedback request   Dan
• [Asterisk-Users] Spinlock with ZAPTEL   Lee Archer
• [Asterisk-Users] AVM CAPI INSTALLATION   Armin Schindler
• [Asterisk-Users] Shoutcast Music On Hold problems?   Patrick
• [Asterisk-Users] HowTo start DIAL by a sillent training as W for modems   f6hqz-m at wanadoo.fr
• [Asterisk-Users] HowTo start DIAL by a sillent training as W for modems   Doug Lytle
• [Asterisk-Users] meetme problem   Felix Skwarczynski
• [Asterisk-Users] meetme problem   Felix Skwarczynski
• [Asterisk-Users] AVM CAPI INSTALLATION   sylvain garcia
• [Asterisk-Users] AVM CAPI INSTALLATION   Armin Schindler
• [Asterisk-Users] meetme problem   Felix Skwarczynski
• [Asterisk-Users] AVM CAPI INSTALLATION   sylvain garcia
• [Asterisk-Users] AGI "say number" but in french   Arvanitis Kostas
• [Asterisk-Users] SixTel?   Jason p
• [Asterisk-Users] TDM card and voicemail volume   Adam Robins
• [Asterisk-Users] AVM CAPI INSTALLATION   Elmar Haneke
• [Asterisk-Users] AGI "say number" but in french   David John Walsh
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   harry gaillac
• [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!   Peter Bowyer
• [Asterisk-Users] OT: Good soft-phone on Linux   Zoltan Szecsei
• [Asterisk-Users] Using asterisk as Quality Monitoring Platform?   Aaron Sundman
• [Asterisk-Users] is teliax down?   Mark Musone
• [Asterisk-Users] Using Conferencing and Meetme   Jean-Marc Salsa
• [Asterisk-Users] pbx_extension_helper: No application 'agi'   John Millican
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   John Goerzen
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   Michael Di Martino
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   John Goerzen
• [Asterisk-Users] Livevoip 800 Choppy Audio   Wiley Siler
• [Asterisk-Users] Asterisk & SpanDSP -> problems by sending a fax   Dominik Simon
• [Asterisk-Users] Using Conferencing and Meetme   Vamsi Pottangi
• [Asterisk-Users] HOW TO WRITE ROUTE PATTERNS DIALPLAN   Jay Milk
• [Asterisk-Users] Using Conferencing and Meetme   Vamsi Pottangi
• [Asterisk-Users] help, switch off NOTICE in console   Anatoly Pugachev
• [Asterisk-Users] pbx_extension_helper: No application 'agi'   Jay Milk
• [Asterisk-Users] Using Conferencing and Meetme   Dean Collins
• [Asterisk-Users] Using Conferencing and Meetme   Jean-Marc Salsa
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   Wiley Siler
• [Asterisk-Users] Re: Using asterisk as Quality Monitoring Platform?   Jason Kawakami
• [Asterisk-Users] simultaneus calls?   Damon Estep
• [Asterisk-Users] ClueCon, Vote?   Brian West
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   Andrew Kohlsmith
• [Asterisk-Users] simultaneus calls?   Erdem HAKİ
• [Asterisk-Users] LiveVoip is Bankrupt   steve szmidt
• [Asterisk-Users] How do you handle NAT?   Matthew Boehm
• [Asterisk-Users] Some phone numbers always busy   Moises Silva
• [Asterisk-Users] How do you handle NAT?   Ray Van Dolson
• [Asterisk-Users] MeetMe application in Asterisk V1.07   monty-asterisk at bc.cx
• [Asterisk-Users] How do you handle NAT?   Jean-Michel Hiver
• [Asterisk-Users] RTP session between two end users   Eric Wieling aka ManxPower
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   John Goerzen
• [Asterisk-Users] list Searchability   Wiley Siler
• [Asterisk-Users] Bad Bad Performance; Max 20 Calls on Quad Proc?   Matthew Boehm
• [Asterisk-Users] TDM400   Jared Armstrong
• [Asterisk-Users] Hangup detection on Panasonic KXTD816   Hilton Williams
• [Asterisk-Users] Using asterisk as Quality Monitoring Platform?   Francois Lambert
• [Asterisk-Users] How do you handle NAT?   Matthew Boehm
• [Asterisk-Users] list Searchability   Steve Hanselman
• [Asterisk-Users] Asterisk dies with Meetme   João Amaro
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 183   Nguyen Trung Tin
• [Asterisk-Users] ASTCC not billing   Ade Agbero
• [Asterisk-Users] pbx_extension_helper: No application 'agi'   Moises Silva
• [Asterisk-Users] Unable to connect to remote asterisk   Jason Greene
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   Wiley Siler
• [Asterisk-Users] ASTCC not billing   Ade Agbero
• [Asterisk-Users] Speech driven crm apps   Dean Collins
• [Asterisk-Users] This weeks Developer meeting   Brian West
• [Asterisk-Users] Junghanns 4 port BRI problem   Doug Reid - Stormcorp
• [Asterisk-Users] Revision I Board TDM04b   Mailing List
• [Asterisk-Users] pbx_extension_helper: No application 'agi'   Tom Fielding
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 183   Nguyen Trung Tin
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 181   Nguyen Trung Tin
• [Asterisk-Users] ASTCC not billing   Bernard Cresencia
• [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.   Michael Welter
• [Asterisk-Users] ASTCC not billing   Bernard Cresencia
• [Asterisk-Users] Hangup detection on Panasonic KXTD816   Eric Wieling aka ManxPower
• [Asterisk-Users] DID in Western Canada   Paul Fielding
• [Asterisk-Users] Correction to Janghanns BRI problem   Doug Reid - Stormcorp
• [Asterisk-Users] simultaneus calls?   Bernard Cresencia
• [Asterisk-Users] problems with chan_capi 0.3.5 , divactrl, eicon diva server, and kernel 2.6.10/2.6.12   Luis Vazquez
• [Asterisk-Users] Revision I Board TDM04b   Tim Pushor
• [Asterisk-Users] CPU load about at max when it should be idle.   gw at adcomcorp.com
• [Asterisk-Users] MeetMe application in Asterisk V1.07   Moises Silva
• [Asterisk-Users] This weeks Developer meeting   Dean Collins
• [Asterisk-Users] Junghanns 4 port BRI problem   Kristof Hardy
• [Asterisk-Users] initial setup: problem   Sean Browne (Cardiff IT Support Ltd)
• [Asterisk-Users] LiveVoip is Bankrupt   Walt Reed
• [Asterisk-Users] Shoutcast Music On Hold problems?   hank
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   hank
• [Asterisk-Users] Voice Mail hangup on not messages   David Romero
• [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]   hank
• [Asterisk-Users] How do you handle NAT?   hank
• [Asterisk-Users] How do you handle NAT?   hank
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   Andrew Kohlsmith
• [Asterisk-Users] Voicemail max time length   Johann
• [Asterisk-Users] enabling stun on asterisk?   hank
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   Tzafrir Cohen
• [Asterisk-Users] Asterisk Realtime and ODBC   Tim Lyakhovetskiy
• [Asterisk-Users] pbx_extension_helper: No application 'agi'   Tom Fielding
• [Asterisk-Users] Mitel SX2000 Integration   David Sampson
• [Asterisk-Users] Shoutcast Music On Hold problems?   Huddleston, Robert
• [Asterisk-Users] Revision I Board TDM04b   Steve Totaro
• [Asterisk-Users] Linksys WRT54GP2-NA settings for performance and low bandwidth?   Paul Fielding
• [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.   qrss
• [Asterisk-Users] VoipJet TOS (was Teliax and also LiveVoip)   Wiley Siler
• [Asterisk-Users] HowTo start DIAL by a sillent training as Wfor modems   f6hqz-m at wanadoo.fr
• [Asterisk-Users] Voicemail max time length   Johann
• [Asterisk-Users] Asterisk Realtime and ODBC   Matthew Boehm
• [Asterisk-Users] astcc no billed cost   Ade Agbero
• [Asterisk-Users] How do you handle NAT?   Sebastian Silva
• [Asterisk-Users] Asterisk Realtime and ODBC   Brian West
• [Asterisk-Users] HowTo start DIAL by a sillent training as Wfor modems   Eric Wieling aka ManxPower
• [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.   Michael Welter
• [Asterisk-Users] enabling stun on asterisk?   Wojciech Tryc
• [Asterisk-Users] LiveVoip is Bankrupt - Why this thread   trixter http://www.0xdecafbad.com
• [Asterisk-Users] Linksys WRT54GP2-NA settings for performance and low bandwidth?   Greg Oliver
• [Asterisk-Users] HowTo start DIAL by a sillent training as Wfor modems   Steve Clark
• [Asterisk-Users] HowTo start DIAL by a sillent training as   Doug Lytle
• [Asterisk-Users] TDM04B Echo on Only One Channel   qrss
• [Asterisk-Users] Dial application timeout   Rob Fugina
• [Asterisk-Users] How do you handle NAT?   Andres
• [Asterisk-Users] MeetMe application in Asterisk V1.07   Monty
• [Asterisk-Users] OT: Good soft-phone on Linux   Jesus Mogollon
• [Asterisk-Users] Trying to get *8 call pickup to work   Robert Woodcock
• [Asterisk-Users] Group/Broadcast Voicemail   Chris Stinson
• [Asterisk-Users] Revision I Board TDM04b   Steve Totaro
• [Asterisk-Users] Inter-Tel 8662 configuration problem.   Alejandro Sosa
• [Asterisk-Users] HowTo start DIAL by a sillent training as W for modems   f6hqz-m at wanadoo.fr
• [Asterisk-Users] How do you handle NAT?   Ray Van Dolson
• [Asterisk-Users] TE110P Card   Juan Pablo Abuyeres
• [Asterisk-Users] Voice Mail hangup on not messages   Moises Silva
• [Asterisk-Users] initial setup: problem   Moises Silva
• [Asterisk-Users] OT: Good soft-phone on Linux   John Mylchreest
• [Asterisk-Users] Group/Broadcast Voicemail   Robert Goodyear
• [Asterisk-Users] VoipJet TOS (was Teliax and also LiveVoip)   steve szmidt
• [Asterisk-Users] Asterisk with Lucent TNT echo   Jeremiah Millay
• [Asterisk-Users] BRIstuff/OctoBRI problem: Ring requested on unconfigured channel 255/255 span 5   Bruno.Voigt at ic3s.de
• [Asterisk-Users] HowTo start DIAL by a sillent training as Wfor modems   Andrew Latham
• [Asterisk-Users] Asterisk Realtime and ODBC   Tim Lyakhovetskiy
• [Asterisk-Users] Asterisk Realtime and ODBC   Tim Lyakhovetskiy
• [Asterisk-Users] How do you handle NAT?   Geert Nijpels
• [Asterisk-Users] Shoutcast Music On Hold problems?   hank
• [Asterisk-Users] Asterisk RSS list feeder ready   Sjaak Nabuurs
• [Asterisk-Users] Asterisk rss list feeder ready   Sjaak Nabuurs
• [Asterisk-Users] TDM04B Echo on Only One Channel   Steve Totaro
• [Asterisk-Users] How do you handle NAT?   hank
• [Asterisk-Users] How do you handle NAT?   OMS
• [Asterisk-Users] Asterisk with Lucent TNT echo   Carlos
• [Asterisk-Users] Net2Phone equipment and different VOIP providers   bph
• [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceand low bandwidth?   Paul Fielding
• [Asterisk-Users] Asterisk RSS list feeder ready   Denis Galvão - iSolve
• [Asterisk-Users] SIP Phone Config Generator   Max Clark
• [Asterisk-Users] Newbie Confusion on Call Forward and DBput/DBdel   Jeffrey Starin
• [Asterisk-Users] MeetMe application in Asterisk V1.07   Moises Silva
• [Asterisk-Users] SIP Phone Config Generator   Tarpo, Louie
• [Asterisk-Users] How do you handle NAT?   Matthew Boehm
• [Asterisk-Users] Unable to connect to remote asterisk   beonice
• [Asterisk-Users] TDM04B Echo on Only One Channel   Steve Totaro
• [Asterisk-Users] ASDI Programming through an ATA/SIP device?   C. Hatton Humphrey
• [Asterisk-Users] manager api call number, pause, dial ext   Thomas Miller
• [Asterisk-Users] Red Hat Enterprise 3.0 issue   Federico Alves
• [Asterisk-Users] How do you handle NAT?   Eric Wieling aka ManxPower
• [Asterisk-Users] Livevoip 800 Choppy Audio   Rich Adamson
• [Asterisk-Users] SIP Phone Config Generator   Steve Totaro
• [Asterisk-Users] Re: ERROR[22927]: Failed to create socketpair for player(24, Too many open files).   Yap Teong Eng
• [Asterisk-Users] Revision I Board TDM04b   Rich Adamson
• [Asterisk-Users] Re: [Asterisk-Dev] ClueCon, Vote?   Greg Boehnlein
• [Asterisk-Users] Anyone using SipP to produce RTP load?   Matthew Boehm
• [Asterisk-Users] Asterisk rss list feeder ready   Frank Mayhar
• [Asterisk-Users] FXO/FXS cpu spikes, data loss and ztclock.   qrss
• [Asterisk-Users] ASTCC not billing   Juan Luis Moyano
• [Asterisk-Users] AMP/A@H (asterisk at home) custom incoming routing   Tom Rymes
• [Asterisk-Users] ASTCC not billing   Juan Luis Moyano
• [Asterisk-Users] Re: ERROR[22927]: Failed to create socketpair for player(24, Too many open files).   Wai-Sun Chia
• [Asterisk-Users] AMP/A@H (asterisk at home) custom incoming routing   David Phelan
• [Asterisk-Users] ASTCC not billing   Juan Luis Moyano
• [Asterisk-Users] audiocodes   Joe Murray
• Fw: [Asterisk-Users] Shoutcast Music On Hold problems?   hank
• [Asterisk-Users] Hop-On WIFI Phone MSRP $40   Cory Andrews
• [Asterisk-Users] Linksys WRT54GP2-NA settings for performanceandlow bandwidth?   Marcel van Kaam, Fonetica
• [Asterisk-Users] RTP session between two end users   Erdem HAKİ
• [Asterisk-Users] cheap HFC card on Bristuff vs cheap HFC card on i4l vs Fritz ISDN BRI card on CAPI   Klaus-Peter Junghanns
• [Asterisk-Users] Outgoing Calls   Micko
• [Asterisk-Users] Anyone using SipP to produce RTP load?   Zoa
• [Asterisk-Users] simultaneus calls?   Erdem HAKİ
• [Asterisk-Users] Junghanns 4 port BRI problem   Klaus-Peter Junghanns
• [Asterisk-Users] Correction to Janghanns BRI problem   Klaus-Peter Junghanns
• [Asterisk-Users] OT: Good soft-phone on Linux   Filippo Carone
• [Asterisk-Users] Anyone using SipP to produce RTP load?   tim panton
• [Asterisk-Users] Anyone using SipP to produce RTP load?   tim panton
• [Asterisk-Users] CallerID Bug?   li770426
• [Asterisk-Users] Re: ERROR[22927]: Failed to create socketpairfor player(24, Too many open files).   Yap Teong Eng
• [Asterisk-Users] ASTCC not billing   Bernard Cresencia
• [Asterisk-Users] chan_capi-cm-0.5.3 fixup release   Armin Schindler
• [Asterisk-Users] GSM Hunting   latte lawson
• [Asterisk-Users] problems with chan_capi 0.3.5 , divactrl, eicon diva server, and kernel 2.6.10/2.6.12   Armin Schindler
• [Asterisk-Users] ASTCC not billing   Ade Agbero
• [Asterisk-Users] GSM Hunting   Florian Overkamp
• [Asterisk-Users] Teliax Problems   Malcolm Taylor
• [Asterisk-Users] CAPI and Caller ID name not showing.   louis g
• [Asterisk-Users] App_conference in dial plan?   Mark Benson
• [Asterisk-Users] Recommend against Teliax as primary ITSP   Chris Coulthurst
• [Asterisk-Users] ast_rtp_read: Unknown RTP codec 100 received21 when receiving fax   Joseph
• [Asterisk-Users] Teliax Problems   Mark Musone
• [Asterisk-Users] Teliax Problems   Chris Coulthurst
• [Asterisk-Users] Teliax Problems   Chris Coulthurst
• [Asterisk-Users] Equipment for small office setup   Steve Foy
• [Asterisk-Users] ASTCC not billing   Bernard Cresencia
• [Asterisk-Users] Teliax Problems   Chris Mason (Lists)
• [Asterisk-Users] Teliax Problems   Rich Adamson
• [Asterisk-Users] AMP/A@H (asterisk at home) custom incoming routing   C. Hatton Humphrey
• [Asterisk-Users] Teliax Problems   Robert Webb
• [Asterisk-Users] Shoutcast Music On Hold problems?   Huddleston, Robert
• [Asterisk-Users] Asterisk LAMP Developer   Dominique Kull
• [Asterisk-Users] PRI got event: HDLC Abort (6) on Primary D-channel of span 1   Michael Blood
• [Asterisk-Users] Machine Sizing   Kevin Roche
• [Asterisk-Users] Asterisk and Cisco CallManager Integration   Barney Sowood
• [Asterisk-Users] Problem with Connecting PBX to Asterisk   Karthik Natarajan
• [Asterisk-Users] Machine Sizing   Doug Lytle
• [Asterisk-Users] Revision I Board TDM04b   Andrew Kohlsmith
• [Asterisk-Users] Teliax Problems   Malcolm Taylor
• [Asterisk-Users] CAPI and Caller ID name not showing.   Armin Schindler
• [Asterisk-Users] Anyone noticed Voipjet voice quality problems?   Pedro
• [Asterisk-Users] Hop-On WIFI Phone MSRP $40   William Suffill
• [Asterisk-Users] TDM card and voicemail volume   David Brodbeck
• [Asterisk-Users] App_conference in dial plan?   Mark Benson
• [Asterisk-Users] CAPI and Caller ID name not showing.   Stefan Gofferje
• [Asterisk-Users] TDM card and voicemail volume   David Brodbeck
• [Asterisk-Users] How to fetch a call not in the same callgroup   Kib Eki
• [Asterisk-Users] timeout on incoming PRI call   Günther Starnberger
• [Asterisk-Users] Trying to get *8 call pickup to work   Tony Nichols
• [Asterisk-Users] Asterisk with Lucent TNT echo   Jeremiah Millay
• [Asterisk-Users] PRI got event: HDLC Abort (6) on Primary D-channel of span 1   Tom Hayden
• [Asterisk-Users] CAPI and Caller ID name not showing.   Christian Händel
• [Asterisk-Users] Asterisk and Cisco CallManager Integration   Greg Oliver
• [Asterisk-Users] Play an announcement to the CALLING party   Stefan Gofferje
• [Asterisk-Users] ASTCC not billing   Ade Agbero
• [Asterisk-Users] audiocodes   Dana Olson
• [Asterisk-Users] Problems with OR Logic in the GotoIf Statement   Keith O'Brien
• [Asterisk-Users] timeout on incoming PRI call   Alexander Lopez
• [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlow bandwidth?   Paul Fielding
• [Asterisk-Users] Teliax Problems   Wiley Siler
• [Asterisk-Users] Teliax Problems   hank
• [Asterisk-Users] Shoutcast Music On Hold problems?   hank
• [Asterisk-Users] Asterisk with Lucent TNT echo   Patrick
• [Asterisk-Users] Trying to get *8 call pickup to work   Brian West
• [Asterisk-Users] OrderlyQ installations?   Jason Kawakami
• [Asterisk-Users] TDM card and voicemail volume   Patrick
• [Asterisk-Users] Teliax Problems   asterisk at txpe.net
• [Asterisk-Users] Problems with OR Logic in the GotoIf Statement   Damon Estep
• [Asterisk-Users] Play an announcement to the CALLING party   Alexander Lopez
• [Asterisk-Users] Can't bridge between h323 and sip calls   Alex Vishnev
• [Asterisk-Users] Equipment for small office setup   Wilson Pickett
• [Asterisk-Users] Problems with OR Logic in the GotoIf Statement   Giorgio Incantalupo
• [Asterisk-Users] Setting Caller ID after Dial   Chee Foong Chiew
• [Asterisk-Users] Teliax Problems   Rick Baranowski
• [Asterisk-Users] Problems with OR Logic in the GotoIf Statement   Giorgio Incantalupo
• [Asterisk-Users] CAPI and Caller ID name not showing.   Armin Schindler
• [Asterisk-Users] Problem with Connecting PBX to Asterisk   qrss
• [Asterisk-Users] Problems with OR Logic in the GotoIf Statement   Doug Lytle
• [Asterisk-Users] Teliax Problems   Malcolm Taylor
• [Asterisk-Users] Problem with Connecting PBX to Asterisk   Karthik Natarajan
• [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlowbandwidth?   Marcel van Kaam, Fonetica
• [Asterisk-Users] Red Hat Enterprise 3.0 issue   Carlos
• [Asterisk-Users] OT: Good soft-phone on Linux   Seth Remington
• [Asterisk-Users] Teliax Problems   Chris Coulthurst
• [Asterisk-Users] ASTCC not billing   Ade Agbero
• [Asterisk-Users] OrderlyQ installations?   Jason Becker
• [Asterisk-Users] Setting Caller ID after Dial   Bryce Chidester
• [Asterisk-Users] Setting Caller ID after Dial   Eric Wieling aka ManxPower
• [Asterisk-Users] Linksys WRT54GP2-NA settings forperformanceandlowbandwidth?   Greg Oliver
• [Asterisk-Users] dtmfmode=inband still broken in *-1.0.7   Joseph
• [Asterisk-Users] How do you handle NAT?   C F
• [Asterisk-Users] Asterisk with Lucent TNT echo   Jeremiah Millay
• [Asterisk-Users] Sangoma and quad card hang up problems   mobilpete
• [Asterisk-Users] Setting Caller ID after Dial   Mark Johnson
• [Asterisk-Users] (no subject)   Giordano Grandis
• [Asterisk-Users] Setting Caller ID after Dial   Mark Johnson
• [Asterisk-Users] Setting Caller ID after Dial   Chris A. Icide
• [Asterisk-Users] Setting Caller ID after Dial   Huddleston, Robert
• [Asterisk-Users] Music oh hold   Giordano Grandis
• [Asterisk-Users] Setting Caller ID after Dial   Eric Wieling aka ManxPower
• [Asterisk-Users] Setting Caller ID after Dial   Matt Fredrickson
• Fw: [Asterisk-Users] Shoutcast Music On Hold problems?   hank
• [Asterisk-Users] Sangoma and quad card hang up problems   mattf
• [Asterisk-Users] Setting Caller ID after Dial   Steve Kennedy
• [Asterisk-Users] Asterisk with Lucent TNT echo   Leon Sun
• [Asterisk-Users] Music oh hold   Joseph
• [Asterisk-Users] Re: Recommend against Teliax as primary ITSP   Adam Megacz
• [Asterisk-Users] Polycom SoundPoint 501 Problem   Craig Bruenderman
• [Asterisk-Users] Setting Caller ID after Dial   Andrew Kohlsmith
• [Asterisk-Users] OT: Good soft-phone on Linux   Oliver Rath
• [Asterisk-Users] Setting Caller ID after Dial   Eric Wieling aka ManxPower
• [Asterisk-Users] PRI: XXX Missing handling for mandatory IE 12 (cs0, Connected Number) XXX   Paul Belanger
• [Asterisk-Users] Setting Caller ID after Dial   Rich Adamson
• [Asterisk-Users] Dial ZAP Problem   Barton Fisher
• [Asterisk-Users] Re: Recommend against Teliax as primary ITSP   Ivan Fetch
• [Asterisk-Users] Setting Caller ID after Dial   Steve Kennedy
• [Asterisk-Users] AMP or Asterisk   Dave Morrow
• [Asterisk-Users] Welcome   Jason Sommers
• [Asterisk-Users] Multiple Timezones with Asterisk   Max Clark
• [Asterisk-Users] How do you handle NAT?   Samy Antoun
• [Asterisk-Users] Equipment for small office setup   Tom Hayden
• [Asterisk-Users] Asterisk LAMP Developer   Jay Milk
• [Asterisk-Users] Setting Caller ID after Dial   Matt
• [Asterisk-Users] Extension Matching.   Chris Modesitt
• [Asterisk-Users] Extension Matching.   Alexander Lopez
• [Asterisk-Users] Asterisk LAMP Developer   William Boehlke
• [Asterisk-Users] Extension Matching.   Doug Lytle
• [Asterisk-Users] Extension Matching.   Aaron Daniel
• [Asterisk-Users] Music oh hold   Chris Stinson
• [Asterisk-Users] Extension Matching.   Chris Modesitt
• [Asterisk-Users] Polycom SoundPoint 501 Problem   Chris Coulthurst
• [Asterisk-Users] Extension Matching.   Chris Modesitt
• [Asterisk-Users] hidecallerid on analog line   chawki hammoud
• [Asterisk-Users] Extension Matching.   Alexander Lopez
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Ade Agbero
• [Asterisk-Users] hidecallerid on analog line   Flu
• [Asterisk-Users] Re: Polycom SoundPoint 501 Problem   Noah Miller
• [Asterisk-Users] Extension Matching.   Alexander Lopez
• [Asterisk-Users] Multiple Timezones with Asterisk   Mark Willis
• [Asterisk-Users] Parking Position   Kelly Opal
• [Asterisk-Users] hidecallerid on analog line   Robert Webb
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Julius Igugu
• [Asterisk-Users] Extension Matching.   Chris Modesitt
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Juan Luis Moyano
• [Asterisk-Users] Re: timeout on incoming PRI call   Günther Starnberger
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   Mike
• [Bulk] Re: [Asterisk-Users] hidecallerid on analog line   Bernard Cresencia
• [Asterisk-Users] Multiple Timezones with Asterisk   hank
• [Asterisk-Users] Problems connecting to and from my Asterisk server :(   Leo Burd
• [Asterisk-Users] Multiple Timezones with Asterisk   Max Clark
• [Asterisk-Users] New Asterisk documentation   harry gaillac
• [Asterisk-Users] Extension Matching.   Jay Milk
• [Asterisk-Users] help with spandsp   Theodore Cekan
• [Asterisk-Users] UK SIP Provider   Steve Foy
• [Asterisk-Users] New Asterisk documentation   Robert Goodyear
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Darren Wiebe
• [Asterisk-Users] Setting Caller ID after Dial   C F
• [Asterisk-Users] Problems connecting to and from my Asterisk server :(   Moises Silva
• [Asterisk-Users] Server hanging   Chris Mason (Lists)
• [Asterisk-Users] Asterisk LAMP Developer   El Flynn
• [Asterisk-Users] Access asterisk features from analog phone during call   Michael Stahl
• [Asterisk-Users] AMP or Asterisk   El Flynn
• [Asterisk-Users] Problems with OR Logic in the GotoIf Statement   Keith O'Brien
• [Asterisk-Users] Kind of Computer to use   Dovid B. Asterisk Users
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   David Shaw
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   David Shaw
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   David Shaw
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   JD Austin
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   David Phelan
• [Asterisk-Users] SIP NOTIFY message   Leon Sun
• [Asterisk-Users] Setting Caller ID after Dial   Chee Foong Chiew
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Darren Wiebe
• [Asterisk-Users] Setting Caller ID after Dial   Chee Foong Chiew
• [Asterisk-Users] ASTCC Issues - Resolved   Darren Wiebe
• [Asterisk-Users] Kind of Computer to use   Sean Cook
• [Asterisk-Users] Kind of Computer to use   Tomas Florian
• [Asterisk-Users] Setting Caller ID after Dial   Mark Johnson
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   hank
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   hank
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   Chris Mason
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   Dean Collins
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   Darren Wiebe
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 11, Issue 198   Michael Stahl
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Darren Wiebe
• [Asterisk-Users] Quality of provider: VocTel   Michael Stahl
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   hank
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   hank
• [Asterisk-Users] Problems with zaptel and voice prompts/voicemail   Jeremy McDermond
• [Asterisk-Users] X100P connected as extension to Panasonic 616 EASA-PHONE   Guillermo Salas M
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   hank
• [Asterisk-Users] X100P connected as extension to Panasonic 616EASA-PHONE   Terry H. Gilsenan
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Juan Luis Moyano
• [Asterisk-Users] X100P connected as extension to Panasonic 616 EASA-PHONE   Jamie Carl
• [Asterisk-Users] failed IAX, SIP registration - email notification   Joseph
• [Asterisk-Users] Asterisk/SER/Call Manager   Dinesh
• [Asterisk-Users] Calls Dropping   Callum McGillivray
• [Asterisk-Users] Can't build cdr_addon_mysql.   Chris Mason (Lists)
• [Asterisk-Users] New Asterisk documentation   Peter Bowyer
• [Asterisk-Users] Can't build cdr_addon_mysql.   Marcel van Kaam, Fonetica
• [Asterisk-Users] Can't build cdr_addon_mysql.   Chris Mason (Lists)
• [Asterisk-Users] Can't build cdr_addon_mysql.   Marcel van Kaam, Fonetica
• [Asterisk-Users] X100P connected as extension to Panasonic 616EASA-PHONE   Terry H. Gilsenan
• [Asterisk-Users] GnuGK and Asterisk   Ronald_Wiplinger
• [Asterisk-Users] Welcome   Jean-Michel Hiver
• [Asterisk-Users] Sipura 3k answers then immediate busy signal   Jane Reeder
• R: [Asterisk-Users] Music oh hold   Giordano Grandis
• [Asterisk-Users] Anyone using SipP to produce RTP load?   Dinesh Nair
• [Asterisk-Users] Trying to get *8 call pickup to work   Klaus-Peter Junghanns
• R: [Asterisk-Users] Music oh hold   Giordano Grandis
• [Asterisk-Users] AMP - recording call   Alexis F.
• [Asterisk-Users] Music oh hold   Marcel van Kaam, Fonetica
• [Asterisk-Users] Problems with zaptel and voice prompts/voicemail   Chris Coulthurst
• [Asterisk-Users] Do any ITSPs support Speex?   Brian Capouch
• [Asterisk-Users] AMP - recording call   hank
• [Asterisk-Users] Resolving groupcalls   Martin Czarnowski
• [Asterisk-Users] hidecallerid on analog line   chawki hammoud
• [Asterisk-Users] How to Playback a file continuously during conversation?   Jean-Hugues ROBERT
• [Asterisk-Users] hidecallerid on analog line   chawki hammoud
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Ade Agbero
• [Asterisk-Users] hidecallerid on analog line   Steve Hanselman
• [Asterisk-Users] hidecallerid on analog line   chawki hammoud
• [Asterisk-Users] AMP or Asterisk   Scott Kamp
• [Asterisk-Users] Resolving groupcalls   Tzafrir Cohen
• [Asterisk-Users] Resolving groupcalls   Chris Coulthurst
• [Asterisk-Users] Resolving groupcalls   Chris Coulthurst
• [Asterisk-Users] New Asterisk documentation   Jeff Heath
• [Asterisk-Users] Getting FOP working with ICD?   Axel Pache
• [Asterisk-Users] Fw: Multiple Quad Bri card   tonini.massimo at tiscali.it
• [Asterisk-Users] Voicemail => SMS   Mark Charlton
• [Asterisk-Users] UK SIP Provider   Mark Benson
• [Asterisk-Users] Extension Matching.   Eric Wieling aka ManxPower
• [Asterisk-Users] problems in dialing in routes patterns   wassim darwish
• [Asterisk-Users] SIP NOTIFY message   Steve Blair
• [Asterisk-Users] voice mail problem   Betül Gözlükoğlu
• [Asterisk-Users] Setting Caller ID after Dial   Eric Wieling aka ManxPower
• [Asterisk-Users] Master.csv and MYSQL   Giorgio Incantalupo
• [Asterisk-Users] Dial ZAP Problem   Eric Wieling aka ManxPower
• [Asterisk-Users] voice mail problem   Yair Hakak
• [Asterisk-Users] PRI got event: HDLC Abort (6) on Primary D-channel of span 1   Eric Wieling aka ManxPower
• [Asterisk-Users] New Asterisk documentation   Zoa
• [Asterisk-Users] Music oh hold   Adam Robins
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   Dean Collins
• [Asterisk-Users] AMP - recording call   Dean Collins
• [Asterisk-Users] [Asterisk-Dev] C Code of Asterisk   Bharat M. Sarvan
• [Asterisk-Users] [Asterisk-Dev] C Code of Asterisk   Bharat M. Sarvan
• [Asterisk-Users] voice mail problem   Erdem HAKİ
• [Asterisk-Users] Voicemail => SMS   Mike Dent
• [Asterisk-Users] Master.csv and MYSQL   Dave Cotton
• [Asterisk-Users] Voicemail => SMS   Mike Dent
• [Asterisk-Users] Failover question   Mohamed A. Gombolaty
• [Asterisk-Users] Voicemail => SMS   Dean Collins
• [Asterisk-Users] X100P connected as extension to Panasonic 616EASA-PHONE   Guillermo Salas M
• [Asterisk-Users] voice mail problem   Betül Gözlükoğlu
• [Asterisk-Users] Asterisk failover solution   Mohamed A. Gombolaty
• [Asterisk-Users] Developing an application in Asterisk.   Bharat M. Sarvan
• [Asterisk-Users] Echo problem   pellegrini at frameweb.it
• [Asterisk-Users] Re: AMP - recording call   Alexis F.
• [Asterisk-Users] [Asterisk-Dev] Developing an Application in Asterisk   Bharat M. Sarvan
• [Asterisk-Users] Setting Caller ID after Dial   Steve Hanselman
• [Asterisk-Users] Developing an Application in Asterisk   Eric Wieling aka ManxPower
• [Asterisk-Users] Developing an Application in Asterisk   Seth Remington
• [Asterisk-Users] SIP NOTIFY message   Christian Hiller
• [Asterisk-Users] Daily Asterisk News   Matt Riddell
• R: [Asterisk-Users] Music oh hold   Giordano Grandis
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Darren Wiebe
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Darren Wiebe
• [Asterisk-Users] [Fwd: Asterisk Balancing solution]   Mohamed A. Gombolaty
• [Asterisk-Users] Fw: Multiple Quad Bri card   Emanuele Pucciarelli
• [Asterisk-Users] Music oh hold   Adam Robins
• [Asterisk-Users] Trying to get *8 call pickup to work   Robert Woodcock
• [Asterisk-Users] Cisco Voip Question   Brian C. Fertig
• [Asterisk-Users] Logrotate   Brian C. Fertig
• [Asterisk-Users] [Fwd: Asterisk Balancing solution]   Damon Estep
• [Asterisk-Users] Logrotate   Geoff Manning
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Ade Agbero
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Ade Agbero
• [Asterisk-Users] [Fwd: Asterisk Balancing solution]   Joshua Colp
• [Asterisk-Users] Asterisk failover solution   Michael Stahl
• [Asterisk-Users] Logrotate   Hilton Williams
• [Asterisk-Users] [Fwd: Asterisk Balancing solution]   Matthew Boehm
• [Asterisk-Users] Logrotate   Eric Wieling aka ManxPower
• [Asterisk-Users] Audiocodes MP-1xx cheat sheet   John Sundberg
• [Asterisk-Users] X100P connected as extension to Panasonic 616EASA-PHONE   John Novack
• [Asterisk-Users] Logrotate   Brian C. Fertig
• [Asterisk-Users] Asterisk failover solution   Vamsi Pottangi
• [Asterisk-Users] Logrotate   Geoff Manning
• [Asterisk-Users] Cisco Voip Question   Greg Oliver
• [Asterisk-Users] [Fwd: Asterisk Balancing solution]   Jay Milk
• [Asterisk-Users] Developing an application in Asterisk.   Moises Silva
• [Asterisk-Users] Logrotate   Brian C. Fertig
• [Asterisk-Users] Voicemail => SMS   Mark Charlton
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 202   Nguyen Trung Tin
• [Asterisk-Users] Logrotate   Dave Cotton
• [Asterisk-Users] voice mail problem   andrew matthews
• [Asterisk-Users] Asterisk failover solution   Deon
• [Asterisk-Users] Can't bridge between h323 and sip calls   Ronald Wiplinger
• [Asterisk-Users] Failover question   Joseph
• [Asterisk-Users] AMP - recording call   hank
• [Asterisk-Users] AMP - recording call   hank
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   hank
• [Asterisk-Users] Asterisk@Home Ver 1.2 Whats new?   hank
• [Asterisk-Users] X100P connected as extension to Panasonic 616EASA-PHONE   Guillermo Salas M
• [Asterisk-Users] Asterisk failover solution   Chris A. Icide
• [Asterisk-Users] [Fwd: Asterisk Balancing solution]   Chris A. Icide
• [Asterisk-Users] Re: Logrotate   Tony Mountifield
• [Asterisk-Users] GUI that supports virtual PBX's/users   Deon
• [Asterisk-Users] Strange dropped calls   Guido Hecken
• [Asterisk-Users] X100P connected as extension to Panasonic 616EASA-PHONE   Guillermo Salas M
• [Asterisk-Users] voice mail problem   Doug Lytle
• [Asterisk-Users] Pickup pin   Ronald Wiplinger
• [Asterisk-Users] Trying to do very simple Zaptel Config. NO LUCK!   David
• [Asterisk-Users] Failover question   Boris Bakchiev
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Juan Luis Moyano
• [Bulk] Re: [Asterisk-Users] Asterisk failover solution   Bernard Cresencia
• [Asterisk-Users] Failover question   Michiel van Baak
• [Asterisk-Users] Failover question   Aaron Daniel
• [Asterisk-Users] GUI that supports virtual PBX's/users   Chris A. Icide
• [Asterisk-Users] Quality of provider: VocTel   Brian West
• [Asterisk-Users] GUI that supports virtual PBX's/users   Rick Baranowski
• [Asterisk-Users] ser --> sip.conf --->extensions.conf, variable context   Iqbal
• [Asterisk-Users] Getting FOP working with ICD?   Nicolás Gudiño
• [Asterisk-Users] Failover question   John Cianfarani
• [Asterisk-Users] GUI that supports virtual PBX's/users   Francois Lambert
• [Asterisk-Users] passing through MWI info from SBC   Mike Myers
• [Asterisk-Users] spandsp fax out fails   Bob Goddard
• [Asterisk-Users] Asterisk & mp3 playback while dialing   Dimitris Kouimintzis
• [Asterisk-Users] Getting FOP working with ICD?   TC
• [Asterisk-Users] callprogress and queues   Daniel Daley
• [Asterisk-Users] Failover question   JD Austin
• [Asterisk-Users] Failover question   Greg Oliver
• [Asterisk-Users] Failover question   Huddleston, Robert
• [Asterisk-Users] Getting FOP working with ICD?   TC
• [Asterisk-Users] Quality of provider: VocTel   Wiley Siler
• [Asterisk-Users] Asterisk failover solution   John Cianfarani
• [Asterisk-Users] Pickup pin   Moises Silva
• [Asterisk-Users] Trying to do very simple Zaptel Config. NO LUCK!   Robert Webb
• [Asterisk-Users] Trying to do very simple Zaptel Config. NO LUCK!   Moises Silva
• [Asterisk-Users] RE: Trying to do very simple Zaptel Config. NO LUCK!   Jason Kawakami
• [Asterisk-Users] Dial Option A(file.gsm)   Jorge Alayon
• [Asterisk-Users] Dial Option A(file.gsm)   Eric Wieling aka ManxPower
• [Asterisk-Users] Gizmo: Skype done right?   Jerry Glomph Black
• [Asterisk-Users] wi-fi phone advice   chawki hammoud
• [Asterisk-Users] X100P connected as extension to Panasonic616EASA-PHONE   Terry H. Gilsenan
• [Asterisk-Users] wi-fi phone advice   Kanuri, Seshu (Company IT)
• [Asterisk-Users] Flash and zap and # key   Jim Archer
• [Asterisk-Users] Trying to do very simple Zaptel Config. NO LUCK!   David
• [Asterisk-Users] Gizmo: Skype done right?   Matt Fredrickson
• [Asterisk-Users] Dial Option A(file.gsm)   Jorge Alayon
• [Asterisk-Users] Problems with zaptel and voice prompts/voicemail   Jeremy McDermond
• [Asterisk-Users] Failover question   Michiel van Baak
• [Asterisk-Users] New Setup with Analog Phone lines   Chris Gamble
• [Asterisk-Users] X100P connected as extension to Panasonic616EASA-PHONE   John Novack
• [Asterisk-Users] New Setup with Analog Phone lines   Bryce Chidester
• [Asterisk-Users] Cell Phones reporting internation calls   MF Hulber
• [Asterisk-Users] [Fwd: Asterisk Balancing solution]   Erik Espinoza
• [Asterisk-Users] New Setup with Analog Phone lines   Chris Gamble
• [Asterisk-Users] Computer to use   Dovid B. Asterisk Users
• [Asterisk-Users] Gizmo: Skype done right?   Erik Espinoza
• [Asterisk-Users] [Fwd: Asterisk Balancing solution]   Michiel van Baak
• [Asterisk-Users] Gizmo: Skype done right?   Michiel van Baak
• [Asterisk-Users] New Setup with Analog Phone lines   Bryce Chidester
• [Asterisk-Users] Gizmo: Skype done right?   hank
• [Asterisk-Users] New Setup with Analog Phone lines   Jay Milk
• [Asterisk-Users] Quality of provider: VocTel   Patrick
• [Asterisk-Users] Gizmo: Skype done right?   Robert Webb
• [Asterisk-Users] spandsp fax out fails   David Romero
• [Asterisk-Users] Gizmo: Skype done right?   hank
• [Asterisk-Users] Computer to use   Terry H. Gilsenan
• [Asterisk-Users] Linux Firewall Question   OMS
• [Asterisk-Users] HELP HELP HELP NEEDED WITH ASTCC   Darren Wiebe
• [Asterisk-Users] Linux Firewall Question   Terry H. Gilsenan
• [Asterisk-Users] X100P connected as extensionto Panasonic616EASA-PHONE   Terry H. Gilsenan
• [Asterisk-Users] passing through MWI info from SBC   gw at adcomcorp.com
• [Asterisk-Users] chan_oh323.c not compiling (cid not found)   Adeel -31
• [Asterisk-Users] Gizmo: Skype done right?   Erik Espinoza
• [Asterisk-Users] Computer to use   Robert Goodyear
• [Asterisk-Users] Computer to use   Guillermo Salas M
• [Asterisk-Users] X100P connected as extensionto Panasonic616EASA-PHONE   John Novack
• [Asterisk-Users] X100P connectedas extensionto Panasonic616EASA-PHONE   Terry H. Gilsenan
• [Asterisk-Users] No BUSY on PRI   Rod Bacon
• [Asterisk-Users] multiples broadvoice lines   Erick Johnson
• [Asterisk-Users] No BUSY on PRI   Andrew Kohlsmith
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 181   arun
• [Asterisk-Users] Do any ITSPs support Speex?   Wilson Pickett
• [Asterisk-Users] UK SIP Provider   Wilson Pickett
• [Asterisk-Users] Linux Firewall Question   Anand S. Katti
• [Asterisk-Users] problems in dialing in routes patterns   Wilson Pickett
• [Asterisk-Users] Voicemail => SMS   Wilson Pickett
• [Asterisk-Users] Flash Zap Channel   Tom Rymes
• [Asterisk-Users] Flash and zap and # key   Wilson Pickett
• [Asterisk-Users] IAX DTMF Problem...   Mark Edwards
• [Asterisk-Users] Outbound answer on TDM400P   Me
• [Asterisk-Users] Linux Firewall Question   Tzafrir Cohen
• [Asterisk-Users] Voicemail => SMS   snacktime
• [Asterisk-Users] Caller ID problem..   Sukardi Shahdan
• [Asterisk-Users] Got this error after my installation when i do ztcfg -vv   Ian Bert Tusil