December 2005 Archives by date

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Starting: Thu Dec 1 00:05:21 MST 2005
Ending: Sat Dec 31 21:47:59 MST 2005
Messages: 3834

• [Asterisk-Users] FW: CDR issues   JP Carballo
• [Asterisk-Users] Asterisk doesn't start   Dinesh Nair
• [Asterisk-Users] App_rxfax problem   René Enskat [Teamware GmbH]
• [Asterisk-Users] prepaid application   scott
• [Asterisk-Users] Call transfer with voicemail password   Giovanni Miano
• [Asterisk-Users] Motherboard choice for asterisk?   Giovanni Miano
• [Asterisk-Users] Voice Mail   Giovanni Miano
• [Asterisk-Users] optipoint 410 and MWI   Wolfgang Lumpp
• [Asterisk-Users] Compiling Asterisk 1.2 from Source on Debian Sarge- Problems   Marcus Deluigi (intern)
• [Asterisk-Users] unable to make calls out using ast@home   ram
• [Asterisk-Users] unable to make calls out using ast@home   Giovanni Miano
• [Asterisk-Users] unable to make calls out using ast@home   ram
• [Asterisk-Users] Compiling Asterisk 1.2 from Source on Debian Sarge- Problems   gincantalupo
• [Asterisk-Users] unable to make calls out using ast@home   Giovanni Miano
• [Asterisk-Users] cannot dial on console on asterisk 1.2   jonny hashem
• [Asterisk-Users] unable to make calls out using ast@home   ram
• [Asterisk-Users] unable to make calls out using ast@home   Giovanni Miano
• [Asterisk-Users] 1.2.0 PRI dropping calls occasionally...   Steve Davies
• [Asterisk-Users] unable to make calls out using ast@home   ram
• [Asterisk-Users] unable to make calls out using ast@home   Giovanni Miano
• [Asterisk-Users] Codec Problem   Code Lover
• [Asterisk-Users] Compiling Asterisk 1.2 from Source on   Hagen Rode
• [Asterisk-Users] Codec Problem   Elmar Haneke
• [Asterisk-Users] Re: pbx or asterisk?   Alejandro Vargas
• [Asterisk-Users] MGCP problem when through internet   Alejandro Vargas
• [Asterisk-Users] two sip phone communication using asterisk server   Alejandro Vargas
• [Asterisk-Users] Re: Newbie question   vivek at staff.ownmail.com
• [Asterisk-Users] Re: pbx or asterisk?   Pablo Allietti
• [Asterisk-Users] cannot dial on console on asterisk 1.2   Steve Totaro
• [Asterisk-Users] Re: MGCP problem when through internet   Alejandro Vargas
• [Asterisk-Users] mail2fax and fax2mail   Rosario Pingaro
• [Asterisk-Users] Re: US e911 reminder   Matt
• [Asterisk-Users] Error on using queue.   gc
• [Asterisk-Users] show queue in BE   Dov Bigio
• [Asterisk-Users] Error on using queue.   Dov Bigio
• [Asterisk-Users] Problems with auto dialout   Tony Spencer
• [Asterisk-Users] WG: App_rxfax problem   René Enskat [Teamware GmbH]
• [Asterisk-Users] WG: App_rxfax problem   Giovanni Miano
• AW: [Asterisk-Users] WG: App_rxfax problem   René Enskat [Teamware GmbH]
• [Asterisk-Users] WG: App_rxfax problem   Rich Adamson
• [Asterisk-Users] Call transfer error   asterisk183
• [Asterisk-Users] Asterisk cluster and astdb   Bruce Ferrell
• [Asterisk-Users] Blind transfer question   Leif Neland
• AW: [Asterisk-Users] WG: App_rxfax problem   René Enskat [Teamware GmbH]
• [Asterisk-Users] Complete Removal of Asterisk   cp
• [Asterisk-Users] Better transfer   Leif Neland
• [Asterisk-Users] Complete Removal of Asterisk   Moises Silva
• [Asterisk-Users] Better transfer   Patrick
• [Asterisk-Users] Problems with auto dialout   tim panton
• [Asterisk-Users] Complete Removal of Asterisk   Joel Vandal
• [Asterisk-Users] voipbuster   Alejandro Vargas
• [Asterisk-Users] Better transfer   Leif Neland
• [Asterisk-Users] meet me message   Dov Bigio
• [Asterisk-Users] Asterisk Realtime 2 Servers calling each other   Miguel Cavazos
• [Asterisk-Users] Queue calls...   Lenz
• [Asterisk-Users] Better transfer   Patrick
• [Asterisk-Users] Complete Removal of Asterisk   Matt
• [Asterisk-Users] voipbuster   Don Fanning
• [Asterisk-Users] Asterisk and Broadvoice, no incoming voice   Janina Sajka
• [Asterisk-Users] voipbuster   Tony Hoyle
• [Asterisk-Users] Re: pbx or asterisk?   Sean Cook
• [Asterisk-Users] Queue calls...   Trey Blancher
• [Asterisk-Users] Asterisk Perl AGI, bug with stream_file() ?   Benoît Mérouze
• [Asterisk-Users] US e911 reminder   C F
• [Asterisk-Users] US e911 reminder   C F
• [Asterisk-Users] Error on using queue.   gc
• [Asterisk-Users] US e911 reminder   trixter aka Bret McDanel
• [Asterisk-Users] Error on using queue.   Dov Bigio
• [Asterisk-Users] voipbuster   Administrator TOOTAI
• [Asterisk-Users] Re: Codec Problem   Code Lover
• [Asterisk-Users] Asterisk Perl AGI, bug with stream_file() ?   Benoît Mérouze
• [Asterisk-Users] voipbuster   turby at seznam.cz
• [Asterisk-Users] Problem compiling libmfcr2 on FC4   Carlos Chavez
• [Asterisk-Users] iaxmodem   Miguel Soto
• [Asterisk-Users] Altering Incoming CallerID   Hugh L. Johnson
• [Asterisk-Users] Problem compiling libmfcr2 on FC4   Steve Underwood
• [Asterisk-Users] Motherboard choice for asterisk?   Michiel van Baak
• [Asterisk-Users] Re: US e911 reminder   Jan Saell
• [Asterisk-Users] Problem compiling libmfcr2 on FC4   Carlos Chavez
• [Asterisk-Users] voipbuster   turby at seznam.cz
• [Asterisk-Users] prepaid application   Innocent Evil
• [Asterisk-Users] Call Recording   Hans Witvliet
• [Asterisk-Users] Call Recording   Michiel van Baak
• [Asterisk-Users] Altering Incoming CallerID   Tom Hayden
• [Asterisk-Users] Altering Incoming CallerID   Innocent Evil
• [Asterisk-Users] format   Tom Hayden
• [Asterisk-Users] Call Recording   Innocent Evil
• [Asterisk-Users] Altering Incoming CallerID   Hugh L. Johnson
• [Asterisk-Users] Call Recording   Dave Walker
• [Asterisk-Users] Altering Incoming CallerID   Hugh L. Johnson
• [Asterisk-Users] Altering Incoming CallerID   Chris Wade
• [Asterisk-Users] prepaid application   Rusty Shackleford
• [Asterisk-Users] voipbuster   Francesco Peeters
• [Asterisk-Users] prepaid application   Steve Totaro
• [Asterisk-Users] chan_bluetooth and Ericsson/SonyEricsson models   Dan
• [Asterisk-Users] Call Recording   Waldo Rubinstein
• [Asterisk-Users] Sip trunk between Avaya S8700 and Asterisk   Art Luke
• [Asterisk-Users] sixtel   Bill Michaelson
• [Asterisk-Users] Cisco 7970   John Riek
• [Asterisk-Users] chan_sip.c error   John Novack
• [Asterisk-Users] CID text stripped over IAX   Jason T. Nelson
• [Asterisk-Users] default user name and password for a2billing   Goran Donev
• [Asterisk-Users] Call Recording   Henri Herscher
• [Asterisk-Users] Installing files   Vladimir Montealegre
• [Asterisk-Users] sixtel   Hadar Pedhazur
• [Asterisk-Users] Pasting phrases together....   Jan Saell
• [Asterisk-Users] Write to text file in dialplan   Andre Courchesne - Consultant
• [Asterisk-Users] Re: Codec Problem   jonc
• [Asterisk-Users] version 1.2 with chan_bluetooth   Jerry Geis
• [Asterisk-Users] showing the hardware status of an * system   lenz
• [Asterisk-Users] Altering Incoming CallerID   Hugh L. Johnson
• [Asterisk-Users] A worrying article   Wayne Gemmell
• [Asterisk-Users] Write to text file in dialplan   Innocent Evil
• [Asterisk-Users] Re: US e911 reminder   Matt
• [Asterisk-Users] Call Recording   Hans Witvliet
• [Asterisk-Users] sixtel   trixter aka Bret McDanel
• [Asterisk-Users] Shoutcast For MOH with Asterisk 1.2   Daniel Wright
• [Asterisk-Users] Linksys SPA-941 Admin Guide   Cory Andrews
• [Asterisk-Users] Two Phones - Same extension?   Mike McMullen
• [Asterisk-Users] Linksys SPA-941 Admin Guide   John Novack
• [Asterisk-Users] Two Phones - Same extension?   jonc
• [Asterisk-Users] Re: US e911 reminder   tracinet
• [Asterisk-Users] ITSP in a box demo updated   Alistair Cunningham
• [Asterisk-Users] Asterisk as a gateway to Index PBX   Jo Knight
• [Asterisk-Users] Can Asterisk do This?   Goran Donev
• [Asterisk-Users] Can Asterisk do This?   Jonathan Attwood
• [Asterisk-Users] Very Weird problem with MeetMe, SIP, Zap and the combo of the three   Nir Simionovich - CTO
• [Asterisk-Users] Asterisk with ooh323 registers with a cisco gatekeeper but disconnects after 300   Bukoka Budoka
• [Asterisk-Users] Can Asterisk do This?   Alistair Cunningham
• [Asterisk-Users] Transfer problem...   Francesco Peeters
• [Asterisk-Users] config Polycom with both SIP provider and Asterisk   Stewart Nelson
• [Asterisk-Users] Write to text file in dialplan   Mojo with Horan & Company, LLC
• [Asterisk-Users] Two Phones - Same extension?   Jan Saell
• [Asterisk-Users] Two Phones - Same extension?   Jason Becker
• [Asterisk-Users] Two Phones - Same extension?   snacktime
• [Asterisk-Users] Linksys SPA-941 Admin Guide   Kerry Garrison
• [Asterisk-Users] Linksys SPA-941 Admin Guide   Senad Jordanovic
• [Asterisk-Users] prepaid application   Darren Wiebe
• [Asterisk-Users] MeetMe with the V (video) option   Dean Collins
• [Asterisk-Users] Re: sixtel   Stewart Nelson
• [Asterisk-Users] Write to text file in dialplan   Innocent Evil
• [Asterisk-Users] Two Phones - Same extension?   Mike McMullen
• [Asterisk-Users] call center dial plan   Zloty
• [Asterisk-Users] Running asterisk within screen   Marcus Deluigi (intern)
• [Asterisk-Users] call center dial plan   David Phelan
• [Asterisk-Users] Two Phones - Same extension?   Dean Collins
• [Asterisk-Users] Hint: how to include dialplan files from remote systems   John Todd
• [Asterisk-Users] Hint: how to include dialplan files from remotesystems   Alexander Lopez
• [Asterisk-Users] default user name and password for a2billing   ram
• [Asterisk-Users] Sip trunk between Avaya S8700 and Asterisk   BJ Weschke
• [Asterisk-Users] IAX trunking frequency parameter works only on initiator side   Branko Samardzic
• [Asterisk-Users] Running asterisk within screen   Luki
• [Asterisk-Users] Re: IAX Service providers in Australia for unlimited inbound   Ben Dinnerville
• [Asterisk-Users] (no subject)   Lakmal
• [Asterisk-Users] version 1.2 with chan_bluetooth   Dan
• [Asterisk-Users] IAX Configuration   Ishanka Anuradha Ranasooriya
• [Asterisk-Users] (no subject)   P.G.C.K. Nirukshitha
• [Asterisk-Users] Running asterisk within screen   Stefan Reuter
• [Asterisk-Users] DTMF on Planet VIP153   Bohuslav Coufal
• [Asterisk-Users] IAX Configuration   Giovanni Miano
• [Asterisk-Users] (no subject)   Giovanni Miano
• [Asterisk-Users] (no subject)   Giovanni Miano
• [Asterisk-Users] WG: App_rxfax problem   René Enskat [Teamware GmbH]
• [Asterisk-Users] Hint: how to include dialplan files from remotesystems   Giovanni Miano
• [Asterisk-Users] IAX Configuration   ram
• [Asterisk-Users] voipbuster   Alejandro Vargas
• [Asterisk-Users] voipbuster   Alejandro Vargas
• [Asterisk-Users] IAX Configuration   Alejandro Vargas
• [Asterisk-Users] IAX Configuration   Alejandro Vargas
• [Asterisk-Users] Write to text file in dialplan   Tzafrir Cohen
• [Asterisk-Users] Zap Channels   FaberK
• [Asterisk-Users] IAX Configuration   ram
• [Asterisk-Users] Hint: how to include dialplan files from remote systems   Tzafrir Cohen
• [Asterisk-Users] voipbuster   Francesco Peeters
• [Asterisk-Users] Asterisk Users Newsgroup   Tomislav Parčina
• [Asterisk-Users] Zap Channels   Xisco Mateu
• [Asterisk-Users] voipbuster   Francesco Peeters
• [Asterisk-Users] voice problems under 8 concurent calles   Matt
• [Asterisk-Users] Re: Very Weird problem with MeetMe, SIP, Zap and the combo of the three   Tony Mountifield
• [Asterisk-Users] Error on using queue.   Lenz
• [Asterisk-Users] Limiting DID calls   omadon
• [Asterisk-Users] Zap Channels   FaberK
• [Asterisk-Users] AGI Problem   Cyrille Demaret
• [Asterisk-Users] Zap Channels   Tzafrir Cohen
• [Asterisk-Users] Running asterisk within screen   Tzafrir Cohen
• [Asterisk-Users] Zap Channels   FaberK
• [Asterisk-Users] BT - DSS   Simon Faulkner
• [Asterisk-Users] Running asterisk within screen   Gavin Hamill
• [Asterisk-Users] Re: Asterisk Users Newsgroup   Tony Mountifield
• [Asterisk-Users] Asterisk-users   P.G.C.K. Nirukshitha
• [Asterisk-Users] Codec Problem   Code Lover
• [Asterisk-Users] IAX Configuration   Alejandro Vargas
• [Asterisk-Users] AGI Problem   Giovanni Miano
• [Asterisk-Users] Limiting DID calls   Giovanni Miano
• [Asterisk-Users] voice problems under 8 concurent calles   Giovanni Miano
• [Asterisk-Users] RE:how to solve error : cannot find extension context 'from-sip'   Tejas Shah
• [Asterisk-Users] equal priority trunks for balancing   Alejandro Vargas
• [Asterisk-Users] change priority by time   Alejandro Vargas
• [Asterisk-Users] RE:how to solve error : cannot find extension context 'from-sip'   Tom Paseka
• [Asterisk-Users] equal priority trunks for balancing   Alistair Cunningham
• [Asterisk-Users] polycom backlight?   Wilson Pickett
• [Asterisk-Users] sixtel   Steve Totaro
• [Asterisk-Users] AGI Problem   Cyrille Demaret
• [Asterisk-Users] asterisk@home isdn   Alejandro Vargas
• [Asterisk-Users] change priority by time   Daniel Wright
• [Asterisk-Users] Error on using queue.   gc
• [Asterisk-Users] WG: App_rxfax problem   Andrew Furey
• [Asterisk-Users] Soft Phone IP   Vladimir Montealegre
• AW: [Asterisk-Users] WG: App_rxfax problem   René Enskat [Teamware GmbH]
• [Asterisk-Users] version 1.2 with chan_bluetooth   Jerry Geis
• [Asterisk-Users] IP Phones   Vladimir Montealegre
• [Asterisk-Users] Polycom DTMF after connection not working   AR Tarzi
• [Asterisk-Users] voipbuster   Patrick
• [Asterisk-Users] DTMF is choppy on the receive   Don Fanning
• [Asterisk-Users] ISDN card Sirrix.PCI4S0   Tomislav Parčina
• [Asterisk-Users] Re: Asterisk fax   Stefan Tichy
• [Asterisk-Users] voipbuster   Francesco Peeters
• [Asterisk-Users] what is your echo solution   Patrick Fortin
• [Asterisk-Users] asterisk@home isdn   Francesco Peeters
• [Asterisk-Users] IAX Configuration   Alejandro Vargas
• [Asterisk-Users] what is your echo solution   Kristof Hardy
• [Asterisk-Users] Queue and agent transfer   Tamas
• [Asterisk-Users] AGI Problem   Giovanni Miano
• [Asterisk-Users] Queue and agent transfer   James Armstrong
• [Asterisk-Users] Re: Asterisk Users Newsgroup   Steven
• [Asterisk-Users] Queue and agent transfer   Lenz
• [Asterisk-Users] Linksys SPA-941 Admin Guide   Paul Hayes
• [Asterisk-Users] sip invite timeouts   Matthew Simpson
• [Asterisk-Users] sip invite timeouts   Kevin P. Fleming
• [Asterisk-Users] Originate calls but can't receive them on a SIP trunk   Amaury BOSSE
• [Asterisk-Users] dial-out and variable inheritance problems   Tamas
• [Asterisk-Users] WG: App_rxfax problem   Rich Adamson
• [Asterisk-Users] Meetme option 'b'   John Daragon
• [Asterisk-Users] what is your echo solution   Rich Adamson
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 11   Tran Tony
• [Asterisk-Users] Sangoma & Asterisk at home   Jess Coburn
• [Asterisk-Users] Sangoma & Asterisk at home   Saul Diaz
• [Asterisk-Users] Sangoma & Asterisk at home   Jess Coburn
• [Asterisk-Users] Sangoma & Asterisk at home   Rob Lith
• [Asterisk-Users] Re: Meetme option 'b'   Tony Mountifield
• [Asterisk-Users] Music on Hold Error   Dave Morrow
• [Asterisk-Users] what is your echo solution   Jared Armstrong
• [Asterisk-Users] v1.2 and cdr badly written   Kristof Hardy
• [Asterisk-Users] Music on Hold Error   Darrick Hartman
• [Asterisk-Users] Help with a Company or Site for a DEMO. AYUDA con una empresa para una DEMO   Alvaro Parres
• [Asterisk-Users] Re: Sangoma & Asterisk at home   Steven
• [Asterisk-Users] Re: Meetme option 'b'   John Daragon
• [Asterisk-Users] "hint" priority in AEL?   Louis-David Mitterrand
• [Asterisk-Users] WG: App_rxfax problem   Steve Underwood
• [Asterisk-Users] Is a BUG ? Hints and incominglimit   Alvaro Parres
• [Asterisk-Users] "hint" priority in AEL?   Kevin P. Fleming
• [Asterisk-Users] Hint: how to include dialplan files from remote systems   John Todd
• [Asterisk-Users] MeetMe with the V (video) option   Matt Riddell
• [Asterisk-Users] DIAXY to DIAXY problems   Alvaro Parres
• [Asterisk-Users] "hint" priority in AEL?   Louis-David Mitterrand
• [Asterisk-Users] ael questions   Paul
• [Asterisk-Users] sip invite timeouts   John Todd
• [Asterisk-Users] Kernel upgrade causes ztdummy to refuse to run   Chris Bagnall
• [Asterisk-Users] Re: DIAXY to DIAXY problems   Alvaro Parres
• [Asterisk-Users] Sangoma & Asterisk at home   Mike Dent
• [Asterisk-Users] Asterisk as a gateway to Index PBX   Steve Rawlings
• [Asterisk-Users] Failover Registration   Max Clark
• [Asterisk-Users] bluetooth   Jerry Geis
• [Asterisk-Users] Codec Problem   jonc
• [Asterisk-Users] Asterisk 1.2 problems   tneuwert at formos.com
• [Asterisk-Users] AGI Problem   Philipp von Klitzing
• [Asterisk-Users] Can I get to a menu system while in a queue??   Chuck Bunn
• [Asterisk-Users] Disposition failed in Asterisk-1.2.0-stable   tracinet
• [Asterisk-Users] Can I get to a menu system while in a queue??   snacktime
• [Asterisk-Users] Cisco CP-7940G drops time from display   Michael Coburn
• [Asterisk-Users] Asterisk as a gateway to Index PBX   Jo Knight
• [Asterisk-Users] ZapHFC cards not maintaining sync?!   Francesco Peeters
• [Asterisk-Users] Kernel upgrade causes ztdummy to refuse to run   Sean Cook
• Subject: Re: [Asterisk-Users] Linksys SPA-941 Admin Guide   alan
• [Asterisk-Users] ZapHFC cards not maintaining sync?!   Kristof Hardy
• [Asterisk-Users] callback script   wassim darwish
• [Asterisk-Users] Fax Service   Kurth Bemis
• [Asterisk-Users] Asterisk 1.2 problems   jonc
• [Asterisk-Users] Fax Service   Vladimir Montealegre
• [Asterisk-Users] Context confict question??   Chuck Bunn
• [Asterisk-Users] What kind of extension numbers can be used in the exit context of a queue?   Chuck Bunn
• [Asterisk-Users] Asterisk 1.2 problems   tneuwert at formos.com
• [Asterisk-Users] Kernel upgrade causes ztdummy to refuse to run   Chris Bagnall
• [Asterisk-Users] Re: Asterisk 1.2 problems (tneuwert@formos.com)   Jeremiah Millay
• [Asterisk-Users] Fax Service   Rich Adamson
• [Asterisk-Users] ZapHFC cards not maintaining sync?!   Francesco Peeters
• [Asterisk-Users] ZapHFC cards not maintaining sync?!   Francesco Peeters
• [Asterisk-Users] Context confict question??   Andy Kuo
• [Asterisk-Users] calling to asterisk and listening to music (GSM) -->>Anyone, please?????   Mark Edwards
• [Asterisk-Users] Context confict question??   jonc
• [Asterisk-Users] USB/Bluetooth Speakerphones   Michael Graves
• [Asterisk-Users] Fax Service   snacktime
• [Asterisk-Users] Sangoma & Asterisk at home   Jess Coburn
• [Asterisk-Users] Asterisk 1.2 problems   jonc
• [Asterisk-Users] Context confict question??   Chuck Bunn
• [Asterisk-Users] sixtel   JP Carballo
• [Asterisk-Users] IAX Configuration   JP Carballo
• [Asterisk-Users] sixtel   Michael Graves
• [Asterisk-Users] echo canceling algorithm   Joe Pukepail
• [Asterisk-Users] Prodding channel failed   isamar at isamarmaia.org
• [Asterisk-Users] Sangoma & Asterisk at home   Mike Dent
• [Asterisk-Users] Testing Sangoma A2022-SO card with Asterisk 1.2   Mike Dent
• [Asterisk-Users] errors with chan_zap.c when installing asterisk-1.2.0-rc2   Arnaud
• [Asterisk-Users] AGI Problem   Cyrille Demaret
• [Asterisk-Users] echo canceling algorithm   gw at adcomcorp.com
• [Asterisk-Users] Problem, to register an ata on an asterisk   asterisk-ml at hnw-net.de
• [Asterisk-Users] Linksys SPA-841 Missing Calls   Dave Morrow
• [Asterisk-Users] Soft Phone IP   hugolivude
• [Asterisk-Users] chan_blutooth   Jerry Geis
• [Asterisk-Users] Re: Linksys SPA-841 Missing Calls   Wolfgang S. Rupprecht
• [Asterisk-Users] Fax Service   Zac Amsler
• [Asterisk-Users] Re: Linksys SPA-841 Missing Calls   Luki
• [Asterisk-Users] Suggestions on Echo Problems   ram
• [Asterisk-Users] Anyone have experience with SellVoip.net   jglucky at blueware.net
• [Asterisk-Users] callback script   Darren Wiebe
• [Asterisk-Users] Broadband VoIP Startup with Asterisk   jglucky at blueware.net
• [Asterisk-Users] Broadband VoIP Startup with Asterisk   Rusty Dekema
• [Asterisk-Users] Codec Problem   Eric "ManxPower" Wieling
• [Asterisk-Users] Need Good UAE calling rates   ram
• [Asterisk-Users] Re: Linksys SPA-841 Missing Calls   Wolfgang S. Rupprecht
• Fw: Re: Re: [Asterisk-Users] Zaptel errors on Debian   Geo
• [Asterisk-Users] Softphone setup   Colt D. Majkrzak
• [Asterisk-Users] Is a BUG ? Hints and incominglimit   Paradise Dove
• [Asterisk-Users] asterisk@home isdn   Karsten Wemheuer
• [Asterisk-Users] WG: App_rxfax problem   René Enskat [Teamware GmbH]
• [Asterisk-Users] asterisk@home isdn   Francesco Peeters
• [Asterisk-Users] Can't compile chan_zap.c on fresh cvs 1.20 checkout   Remco Barendse
• [Asterisk-Users] Iax2 connection failed   jonny hashem
• [Asterisk-Users] Voip providers with trunked iax2   chawki hammoud
• [Asterisk-Users] IAX Conf Realtime?   Nate Kapi
• [Asterisk-Users] Iax2 connection failed   Administrator TOOTAI
• [Asterisk-Users] Iax2 connection failed   chawki hammoud
• [Asterisk-Users] prepaid application   Scott Pinhorne
• [Asterisk-Users] Iax2 connection failed   Administrator TOOTAI
• [Asterisk-Users] chan_blutooth   Rico -mc- Gloeckner
• [Asterisk-Users] Can't compile chan_zap.c on fresh cvs 1.20 checkout   Andrew Kohlsmith
• [Asterisk-Users] Can't compile chan_zap.c on fresh cvs 1.20 checkout   Remco Barende
• [Asterisk-Users] Can't compile chan_zap.c on fresh cvs 1.20 checkout   Rich Adamson
• [Asterisk-Users] Rates for Asian countries   Amir Aziz
• [Asterisk-Users] No CID Info an TE405P with zaptel 1.2.0   BK
• [Asterisk-Users] WG: App_rxfax problem   Nate Turnbow
• [Asterisk-Users] Fwd: Queue Statistics   Carlos Prieto
• [Asterisk-Users] Can't compile chan_zap.c on fresh cvs 1.20 checkout   Remco Barende
• [Asterisk-Users] ZapHFC cards not maintaining sync?!   Francesco Peeters (Asterisk)
• [Asterisk-Users] Asterisk 1.2 and weird ZAP interface behaviour   Remco Barende
• [Asterisk-Users] Sangoma & Asterisk at home   Tom Rymes
• [Asterisk-Users] chan_blutooth   Ben Higley
• [Asterisk-Users] Asterisk 1.2 and weird ZAP interface behaviour   Begumisa Gerald M
• [Asterisk-Users] Echo!!   Vladimir Montealegre
• [Asterisk-Users] Sangoma & Asterisk at home   Jess Coburn
• [Asterisk-Users] ZapHFC cards not maintaining sync?!   Francesco Peeters (Asterisk)
• [Asterisk-Users] Asterisk 1.2 and weird ZAP interface behaviour   Remco Barende
• [Asterisk-Users] ZapHFC cards not maintaining sync?!   Remco Barende
• [Asterisk-Users] Asterisk 1.2 and weird ZAP interface behaviour   Begumisa Gerald M
• [Asterisk-Users] Linksys SPA-841 Missing Calls   Craig
• [Asterisk-Users] Can I escape queue with a '*'?   Chuck Bunn
• [Asterisk-Users] Extension Manual   Vladimir Montealegre
• [Asterisk-Users] Re: IAX Conf Realtime?   joey murda
• [Asterisk-Users] Order of ports on rear of Sangoma card and pictures in a mini-itx chassis.   Mike Dent
• [Asterisk-Users] Iax2 connection failed   chawki hammoud
• [Asterisk-Users] Extension Manual   Vladimir Montealegre
• [Asterisk-Users] Extension Manual   Yair Hakak
• [Asterisk-Users] Anyone have experience with SellVoip.net   Lists
• [Asterisk-Users] ZapHFC cards not maintaining sync?!   Francesco Peeters (Asterisk)
• [Asterisk-Users] Iax2 connection failed   tim panton
• [Asterisk-Users] Iax2 connection failed   chawki hammoud
• [Asterisk-Users] Call queues, agents with DND status set.   Vladimir S. Blazhkun
• [Asterisk-Users] Extension Manual   Chuck Bunn
• [Asterisk-Users] Sangoma & Asterisk at home   Jess Coburn
• [Asterisk-Users] Adit 600 and Groundstart   Doug Lytle
• [Asterisk-Users] Asterisk 1.2 problems   Waldo Rubinstein
• [Asterisk-Users] Converted mp3 files to raw for musiconhold and still does not work...   Chuck Bunn
• [Asterisk-Users] Converted mp3 files to raw for musiconhold and still does not work...   Bharath
• [Asterisk-Users] polycom backlight?   Script Head
• [Asterisk-Users] Asterisk cluster and astdb   Script Head
• [Asterisk-Users] Broadband VoIP Startup with Asterisk   Chris Mason (Lists)
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 18   hrishikesh shrivastaw
• [Asterisk-Users] Re: missing libpq.so.4   JP Carballo
• [Asterisk-Users] UK Patches for Asterisk 1.2   Vassilis Konstantinou
• Fw: Re: Re: [Asterisk-Users] Zaptel errors on Debian   Tzafrir Cohen
• [Asterisk-Users] UK Patches for Asterisk 1.2   Tzafrir Cohen
• [Asterisk-Users] asterisk@home isdn   Tzafrir Cohen
• [Asterisk-Users] Sipura 3000 Disconnect Singnel   Code Lover
• [Asterisk-Users] UK Patches for Asterisk 1.2   Vassilis Konstantinou
• [Asterisk-Users] Iax2 connection failed   tim panton
• [Asterisk-Users] Call queues, agents with DND status set.   lenz
• [Asterisk-Users] Iax2 connection failed   chawki hammoud
• [Asterisk-Users] replace hard-phone if soft-phone is online   dima
• [Asterisk-Users] replace hard-phone if soft-phone is online   James B. MacLean
• [Asterisk-Users] Iax2 connection failed   tim panton
• [Asterisk-Users] UK Patches for Asterisk 1.2   Tony Hoyle
• [Asterisk-Users] re: Help required on asterisk   Tejas Shah
• [Asterisk-Users] re: Help required on asterisk   Tzafrir Cohen
• [Asterisk-Users] ISDN 2e Cards   Simon Faulkner
• [Asterisk-Users] ISDN 2e Cards   Faris Raouf
• [Asterisk-Users] Why does musiconhold.conf changes require a reboot?   Chuck Bunn
• [Asterisk-Users] New to Asterisk@Home   Dakota
• [Asterisk-Users] RE: how to remove asterisk 1.2 from Red Hat 9   Tejas Shah
• [Asterisk-Users] ISDN 2e Cards   John Daragon
• [Asterisk-Users] ISDN 2e Cards   Avi Miller
• [Asterisk-Users] Failover Registration   Script Head
• [Asterisk-Users] Failover Registration   Eric "ManxPower" Wieling
• [Asterisk-Users] Iax2 connection failed   chawki hammoud
• [Asterisk-Users] Getting started with Asterisk and Aastra 9133i   robertlaferla at comcast.net
• [Asterisk-Users] ISDN 2e Cards   Armin Schindler
• [Asterisk-Users] Sending data over ZAPHFC D-channel?   Francesco Peeters (Asterisk)
• [Asterisk-Users] ISDN 2e Cards   Avi Miller
• [Asterisk-Users] Sipura 3000 Disconnect Singnel   Robert Rozman
• [Asterisk-Users] Failover Registration   Rich Adamson
• [Asterisk-Users] DISA function   Richard Smith
• [Asterisk-Users] UK DID 0208 £1 per month   Sam Tam
• [Asterisk-Users] GSM Gateway / Terminal for sale   Sam Tam
• [Asterisk-Users] New to Asterisk@Home   Kerry Garrison
• [Asterisk-Users] ISDN 2e Cards   Faris Raouf
• [Asterisk-Users] Asterisk 1.2 and weird ZAP interface behaviour   Eric "ManxPower" Wieling
• [Asterisk-Users] Asterisk 1.2 and weird ZAP interface behaviour   Eric "ManxPower" Wieling
• [Asterisk-Users] New to Asterisk@Home   Tom Vile
• [Asterisk-Users] Broadband VoIP Startup with Asterisk   Daniel Wright
• [Asterisk-Users] diax not working properly   amna saleem
• [Asterisk-Users] Connecting 2 Asterisk using SIP   Waldo Rubinstein
• [Asterisk-Users] ZapHFC cards not maintaining sync?!   Remco Barende
• [Asterisk-Users] Sipura 3000 Call waiting on the PSTN line   cp
• [Asterisk-Users] Sipura 3000 Call waiting on the PSTN line   Luki
• [Asterisk-Users] Connecting 2 Asterisk using SIP   Luki
• [Asterisk-Users] Re: sound problem in X-Lite phone with asterisk server   Vipul Patel
• [Asterisk-Users] [Amportal-users] AMP queues, AddQueueMember and 'Wrapup Time'   Adrian Carter
• [Asterisk-Users] voicexml vendors   snacktime
• [Asterisk-Users] Error when compiling asterisk   jourdan lemieux
• [Asterisk-Users] Asterisk 1.2.0 - TE210P - "...Control Frame 15..."   George Vagenas
• [Asterisk-Users] VegaStream 400   scott
• [Asterisk-Users] Re: Sipura 3000 Call waiting on the PSTN line   altus
• [Asterisk-Users] Failover Registration   Daniel Wright
• [Asterisk-Users] ZapHFC cards not maintaining sync?!   Kristof Hardy
• [Asterisk-Users] ZapHFC cards not maintaining sync?!   Francesco Peeters
• [Asterisk-Users] asterisk.h   Tomislav Parčina
• [Asterisk-Users] Chat Lines/ Party Line Solutions for Asterisk   Nate Kapi
• [Asterisk-Users] AGI Problem   Giovanni Miano
• [Asterisk-Users] Connecting 2 Asterisk using SIP   xcel
• [Asterisk-Users] Re: [Asterisk-biz] UK DID 0208 £1 per month   Linus Surguy
• [Asterisk-Users] Iax2 connection failed   tim panton
• [Asterisk-Users] Sound problem   Vipul Patel
• [Asterisk-Users] diax not working properly   Dan
• [Asterisk-Users] (warning) iaxy.bin fails checksum   Benoît Mérouze
• [Asterisk-Users] Calls to DISA over ISDN PRI don't get CONNECT ACKNOLEDGE   Nenad Radosavljevic
• [Asterisk-Users] SIP INVITE with no 'Contact' field and RealTime support.   lokotes
• [Asterisk-Users] New to Asterisk@Home   Dakota
• [Asterisk-Users] Sipura 3000 Call waiting on the PSTN line   Rich Adamson
• [Asterisk-Users] oh323 installation   Code Lover
• [Asterisk-Users] Sipura 3000 Call waiting on the PSTN line   Rich Adamson
• [Asterisk-Users] Sipura 3000 Call waiting on the PSTN line   Rich Adamson
• [Asterisk-Users] Include a variable from another file in config files   Amaury BOSSE
• [Asterisk-Users] Sound problem   Giovanni Miano
• [Asterisk-Users] Include a variable from another file in config files   Giovanni Miano
• [Asterisk-Users] Include a variable from another file in config files   Administrator TOOTAI
• [Asterisk-Users] A@H with a2Billing   ram
• [Asterisk-Users] Re: sound problem in X-Lite phone with asterisk server   Chuck Bunn
• [Asterisk-Users] Linksys SPA-841 Missing Calls   Dave Morrow
• [Asterisk-Users] Streaming MOH   Stojan Sljivic - GDS
• [Asterisk-Users] Problem with a second incoming call on a BRI Zap Channel   David Masure
• [Asterisk-Users] diax not working properly   Time Bandit
• [Asterisk-Users] SNOM and 1.0.9   Justin Carlson
• [Asterisk-Users] Linksys SPA-841 Missing Calls   Rich Adamson
• [Asterisk-Users] SIP INVITE with no 'Contact' field and RealTime support.   Kevin P. Fleming
• [Asterisk-Users] warning message   Patrick Fortin
• [Asterisk-Users] Transfer/take call to/from other phone   Denny Schierz
• [Asterisk-Users] h323 vs oh323   Innocent Evil
• [Asterisk-Users] h323 vs oh323   David Waugh
• [Asterisk-Users] Ambient Modem   Vladimir Montealegre
• [Asterisk-Users] Transfer/take call to/from other phone   Chuck Bunn
• [Asterisk-Users] Connecting 2 Asterisk using SIP   Waldo Rubinstein
• [Asterisk-Users] Connecting 2 Asterisk using SIP   Waldo Rubinstein
• [Asterisk-Users] Restore logging functionality...   Chuck Bunn
• [Asterisk-Users] kernel lockup with Fedora Core 4.0 2.6.14-1.1637   Wade Hampton
• [Asterisk-Users] Restore logging functionality...   Kristof Hardy
• [Asterisk-Users] Getting started with Asterisk and Aastra 9133i   Robert La Ferla
• [Asterisk-Users] Getting started with Asterisk and Aastra 9133i   Robert La Ferla
• [Asterisk-Users] DISA function   Joe Pukepail
• [Asterisk-Users] Getting started with Asterisk and Aastra 9133i   Robert La Ferla
• [Asterisk-Users] Getting started with Asterisk and Aastra 9133i   Pete Barnwell
• [Asterisk-Users] Getting started with Asterisk and Aastra 9133i   Dave Cotton
• [Asterisk-Users] Transfer/take call to/from other phone   Denny Schierz
• [Asterisk-Users] Looking for advice on cell carrier's default "Un avaliable" message   Colin Anderson
• [Asterisk-Users] Looking for advice on cell carrier's default "Un avaliable" message   Joe Pukepail
• [Asterisk-Users] Getting started with Asterisk and Aastra 9133i   Robert La Ferla
• [Asterisk-Users] asterisk won't answer malformed caller id   D. J. Williams
• [Asterisk-Users] Looking for advice on cell carrier's default "Un avaliable" message   Colin Anderson
• [Asterisk-Users] Transfer/take call to/from other phone   James Armstrong
• [Asterisk-Users] Include a variable from another file in configfiles   amaury BOSSE
• [Asterisk-Users] Looking for advice on cell carrier's default "Un avaliable" message   Andrew Kohlsmith
• [Asterisk-Users] Transfer/take call to/from other phone   Chuck Bunn
• solved (Re: [Asterisk-Users] Getting started with Asterisk and Aastra 9133i)   Robert La Ferla
• [Asterisk-Users] Restore logging functionality...   Chuck Bunn
• [Asterisk-Users] Asterisk Queues Tutorial updated...   Matt King
• [Asterisk-Users] Panasonic DBS DISA   Steven
• [Asterisk-Users] Re: Asterisk 1.2 problems (tneuwert@formos.com)   tneuwert at formos.com
• [Asterisk-Users] Looking for advice on cell carrier's default "Un avaliable" message   Colin Anderson
• [Asterisk-Users] Asterisk 1.2 problems   tneuwert at formos.com
• [Asterisk-Users] Looking for advice on cell carrier's default "Un avaliable" message   Andrew Kohlsmith
• [Asterisk-Users] video phones   Jonathan k. Creasy
• [Asterisk-Users] Anyone know anything about the new Linksys One product - does it use Asterisk?   Chuck Bunn
• [Asterisk-Users] DISA function   AR Tarzi
• [Asterisk-Users] Anyone know anything about the new Linksys One product - does it use Asterisk?   Cory Andrews
• [Asterisk-Users] Anyone know anything about the new Linksys One product - does it use Asterisk?   Kerry Garrison
• [Asterisk-Users] Error when compiling asterisk   jourdan lemieux
• [Asterisk-Users] transfers from Polycom 501 involving Sipura 300 and asterisk 1.2   C F
• [Asterisk-Users] Linksys SPA-941 DTMF failure with asterisk v.1.2   tracinet
• [Asterisk-Users] Preventing incoming calls from ringing SIP lines   Paul Redstone
• [Asterisk-Users] h323 vs oh323   isamar at isamarmaia.org
• [Asterisk-Users] Linksys SPA-841 Missing Calls   alan
• [Asterisk-Users] h323 vs oh323   isamar at isamarmaia.org
• [Asterisk-Users] Re: Linksys SPA-941 DTMF failure with asterisk v.1.2   tracinet
• [Asterisk-Users] Asterisk + WiFi Phones   Philip Edelbrock
• [Asterisk-Users] PRI indications.   Adam Rybak
• [Asterisk-Users] Include a variable from another file in configfiles   JP Carballo
• [Asterisk-Users] Best Switch for VOIP Applications   calvis
• [Asterisk-Users] h323 vs oh323   Boris Bakchiev
• [Asterisk-Users] Best Switch for VOIP Applications   Wiley Siler
• [Asterisk-Users] Best Switch for VOIP Applications   calvis
• [Asterisk-Users] h323 vs oh323   Innocent Evil
• [Asterisk-Users] Best Switch for VOIP Applications   Leo Ann Boon
• [Asterisk-Users] Best Switch for VOIP Applications   snacktime
• [Asterisk-Users] Include a variable from another file in configfiles   JP Carballo
• [Asterisk-Users] Best Switch for VOIP Applications   Michiel van Baak
• [Asterisk-Users] Best Switch for VOIP Applications   Wiley Siler
• [Asterisk-Users] Grandstream NTP   Rod Bacon
• [Asterisk-Users] Messages button on a Polycom 501   Brent Bloodworth
• [Asterisk-Users] hierarchical VoIP system   Joao Pereira
• [Asterisk-Users] Best Switch for VOIP Applications   Eric "ManxPower" Wieling
• [Asterisk-Users] Looking for advice on cell carrier's default "Un avaliable" message   Eric "ManxPower" Wieling
• [Asterisk-Users] ADIT 600 T1 with DNIS digits problem   William K. Volkman
• [Asterisk-Users] Messages button on a Polycom 501   Jerry Jones
• [Asterisk-Users] Grandstream NTP   Rod Bacon
• [Asterisk-Users] h323 vs oh323   isamar at isamarmaia.org
• [Asterisk-Users] h323 vs oh323   isamar at isamarmaia.org
• [Asterisk-Users] Messages button on a Polycom 501   Brent Bloodworth
• [Asterisk-Users] Is a BUG ? Hints and incominglimit   Alvaro Parres
• [Asterisk-Users] Best Switch for VOIP Applications   Michael Welter
• [Asterisk-Users] logging performance, important impact?   Moises Silva
• [Asterisk-Users] uip200 phone not work with 1.2   Jerry Geis
• [Asterisk-Users] sipura "Vertical Service Activation Codes"   Wolfgang S. Rupprecht
• [Asterisk-Users] Asterisk on PPC & chan_capi issue   Patrick
• [Asterisk-Users] Best Switch for VOIP Applications   pdhales at optusnet.com.au
• [Asterisk-Users] Preventing incoming calls from ringing SIP lines   James B. MacLean
• [Asterisk-Users] h323 vs oh323   Boris Bakchiev
• [Asterisk-Users] diax not working properly   amna saleem
• [Asterisk-Users] Please help in writing AGI script   Ryan Pagquil
• [Asterisk-Users] 2 leg bridged call not hanging up until both legs hangup   Aaron Bostick
• [Asterisk-Users] Realtime SIP Lookups   Douglas Garstang
• [Asterisk-Users] h323 vs oh323   isamar at isamarmaia.org
• [Asterisk-Users] Messages button on a Polycom 501   Jerry Jones
• [Asterisk-Users] EAGI Audio Capture   ha i
• [Asterisk-Users] Echo cancellation over satellite link   funny guy
• [Asterisk-Users] Echo cancellation over satellite link   BJ Weschke
• [Asterisk-Users] Echo cancellation over satellite link   Jerry Jones
• [Asterisk-Users] Echo cancellation over satellite link   Boris Bakchiev
• [Asterisk-Users] How to restric user to call only specified country   ram
• [Asterisk-Users] Grandstream NTP   Kristof Hardy
• [Asterisk-Users] OH323 user configuration   Code Lover
• [Asterisk-Users] Error when compiling asterisk   Mark Quitoriano
• [Asterisk-Users] logging performance, important impact?   Simone Cittadini
• [Asterisk-Users] How to restric user to call only specified country   Simone Cittadini
• [Asterisk-Users] SNOM Shared line DevState   asterisk
• [Asterisk-Users] zaptel : unresolved symbol zt_unregister ...   Eugene Prokopiev
• [Asterisk-Users] SIP INVITE with no 'Contact' field and RealTime support.   lokotes
• [Asterisk-Users] Asterisk and Video   Dinesh
• [Asterisk-Users] zaptel : unresolved symbol zt_unregister ...   Tzafrir Cohen
• [Asterisk-Users] Include a variable from another file in configfiles   Administrator TOOTAI
• [Asterisk-Users] DISA function   Richard Smith
• [Asterisk-Users] Problem with a second incoming call on a BRI ZapChannel   David Masure
• [Asterisk-Users] RE:Is it possible to install ZAPTEL after installation of Asterisk   Tejas Shah
• [Asterisk-Users] CallParking and chan_capi-cm-0.6   richard Coco
• [Asterisk-Users] RE:Is it possible to install ZAPTEL after installation of Asterisk   Tzafrir Cohen
• [Asterisk-Users] Echo cancellation over satellite link   Randall Prentice
• [Asterisk-Users] Problem with a second incoming call on a BRI ZapChannel   Tzafrir Cohen
• [Asterisk-Users] zaptel : unresolved symbol zt_unregister ...   Eugene Prokopiev
• [Asterisk-Users] Problem with a second incoming call on a BRIZapChannel   David Masure
• [Asterisk-Users] CDR Accounting Problem   René Enskat [Teamware GmbH]
• [Asterisk-Users] IAX2 to SIP   Dijkstra, Roelof
• [Asterisk-Users] diax not working properly   Dan
• [Asterisk-Users] RE: OH323 user configuration   Code Lover
• [Asterisk-Users] Primary D-Channel on span 1 up   Francisco Gutierrez
• [Asterisk-Users] Fax2mail   Andrew Nowrot
• [Asterisk-Users] PRI and Dialed Number   Tobias Ahlander
• [Asterisk-Users] Re: sound problem in X-Lite phone with asterisk server   Steve Davies
• [Asterisk-Users] Preventing incoming calls from ringing SIP lines   Tom Rymes
• [Asterisk-Users] Re: EAGI Audio Capture   ha i
• [Asterisk-Users] SIP Canreinvite   Giordano Grandis
• [Asterisk-Users] Messages button on a Polycom 501   Eric "ManxPower" Wieling
• [Asterisk-Users] Echo cancellation over satellite link   Eric "ManxPower" Wieling
• [Asterisk-Users] Error when compiling asterisk   jourdan lemieux
• [Asterisk-Users] Echo cancellation over satellite link   Andrew Kohlsmith
• [Asterisk-Users] Call Forwarding with Account Code.. can it be done?   Matt
• [Asterisk-Users] SIP INVITE with no 'Contact' field and RealTime support.   Kevin P. Fleming
• [Asterisk-Users] What would prevent logs from being recreated if they are deleted?   Chuck Bunn
• [Asterisk-Users] Dial application "g" option   Peter Krnjevic
• [Asterisk-Users] IAX jitterbuffer and trunking settings between 1.0.9 and 1.2   Chris Bagnall
• [Asterisk-Users] Fax2mail   Colin Anderson
• [Asterisk-Users] IAX jitterbuffer and trunking settings between 1.0.9 and 1.2   Patrick
• [Asterisk-Users] Realtime SIP Lookups   Anish Basu
• [Asterisk-Users] RE: Realtime SIP Lookups   Douglas Garstang
• [Asterisk-Users] Odd DTMF issue over PRI   Steven
• [Asterisk-Users] uip200 phone not work with 1.2   Jason Becker
• [Asterisk-Users] RE: Realtime SIP Lookups   Douglas Garstang
• [Asterisk-Users] Credit Card Terminals   Garrett Smith
• [Asterisk-Users] Connecting 2 Asterisk using SIP   Waldo Rubinstein
• [Asterisk-Users] Messages button on a Polycom 501   Brent Bloodworth
• [Asterisk-Users] Complicated Dialing plan routing   Kristian Larsson
• [Asterisk-Users] Complicated Dialing plan routing   Colin Anderson
• [Asterisk-Users] activex   Vladimir Montealegre
• [Asterisk-Users] activex   Kerry Garrison
• [Asterisk-Users] Streaming MOH   Stojan Sljivic - GDS
• [Asterisk-Users] activex   Vladimir Montealegre
• [Asterisk-Users] E1 and hardware Test.   Juan Salas
• [Asterisk-Users] Streaming MOH   Tzafrir Cohen
• [Asterisk-Users] activex   Bartosz Piec
• [Asterisk-Users] Per Extension Password for Outgoing Routing   Carlos Prieto
• [Asterisk-Users] activex   Vladimir Montealegre
• [Asterisk-Users] can * translate DTMF from rfc2833 to inband?   Damon Estep
• [Asterisk-Users] Messages button on a Polycom 501   Mojo with Horan & Company, LLC
• [Asterisk-Users] Looking for advice on cell carrier's default "Unavaliable" message   Chris Bagnall
• [Asterisk-Users] FAX   turby at seznam.cz
• [Asterisk-Users] Per Extension Password for Outgoing Routing   Innocent Evil
• [Asterisk-Users] snom 320 'retrieve' button   Sean Kennedy
• [Asterisk-Users] Per Extension Password for Outgoing Routing   Joel Vandal
• [Asterisk-Users] VoIPJet issue == No one is available to answer at this time   Cavanna, Richard
• [Asterisk-Users] snom 320 'retrieve' button   Michiel van Baak
• [Asterisk-Users] Per Extension Password for Outgoing Routing   C F
• [Asterisk-Users] FAX   C F
• [Asterisk-Users] Credit Card Terminals   C F
• [Asterisk-Users] SNOM Shared line DevState   Philipp von Klitzing
• [Asterisk-Users] snom 320 'retrieve' button   Colin Anderson
• [Asterisk-Users] Call Forwarding with Account Code.. can it be done?   C F
• [Asterisk-Users] Echo cancellation over satellite link   C F
• [Asterisk-Users] Call Forwarding with Account Code.. can it be done?   Andy Kuo
• [Asterisk-Users] SNOM Shared line DevState   Philipp von Klitzing
• [Asterisk-Users] Re: VoipBuster / Finarea   Wolf N. Paul
• [Asterisk-Users] Per Extension Password for Outgoing Routing   Carlos Prieto
• [Asterisk-Users] Toll-free number on a PRI   Michael Welter
• [Asterisk-Users] IAX or SIP   Ross C
• [Asterisk-Users] Toll-free number on a PRI   Richard Cook
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 7   Wolf N. Paul
• [Asterisk-Users] hierarchical VoIP system   Jan Saell
• [Asterisk-Users] How to setup Connected number   magic.c at gmx.at
• [Asterisk-Users] RE: OH323 user configuration   brett at websmyths.com
• [Asterisk-Users] Complicated Dialing plan routing   'kristian at netatonce.se'
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 7   Vladimir Montealegre
• [Asterisk-Users] Call Forwarding with Account Code.. can it be done?   Matt
• [Asterisk-Users] Messages button on a Polycom 501   Robert Augustyn
• [Asterisk-Users] Call Forwarding with Account Code.. can it be done?   C F
• [Asterisk-Users] Packeteer ? Edgemark ? How to not re-cable ?   A_ Navone
• [Asterisk-Users] can * translate DTMF from rfc2833 to inband?   tracinet
• [Asterisk-Users] Packeteer ? Edgemark ? How to not re-cable ?   Colin Anderson
• [Asterisk-Users] Outgoing fax detection   Andre Courchesne - Consultant
• [Asterisk-Users] EAGI Audio Capture   ha i
• [Asterisk-Users] Packeteer ? Edgemark ? How to not re-cable ?   Stijn Jonker
• [Asterisk-Users] Packeteer ? Edgemark ? How to not re-cable ?   Dean Collins
• [Asterisk-Users] Packeteer ? Edgemark ? How to not re-cable ?   Lawrence Jovellanos
• [Asterisk-Users] [Amportal-users] AMP queues, AddQueueMember and 'Wrapup Time'   pdhales at optusnet.com.au
• [Asterisk-Users] VoIPJet issue == No one is available to answer at this time   David K Parker
• [Asterisk-Users] Call Forwarding with Account Code.. can it be done?   Matt
• [Asterisk-Users] VoIPJet issue == No one is available to answer at this time   Tom Vile
• [Asterisk-Users] VoIPJet issue == No one is available to answer at this time   Matt
• [Asterisk-Users] Re: VoipBuster / Finarea   Tony Hoyle
• [Asterisk-Users] Re: VoipBuster / Finarea   Francesco Peeters
• [Asterisk-Users] ATA Registration problems   Jerry Jones
• [Asterisk-Users] Snom monitoring of extensions not working   Ivan Lopez
• [Asterisk-Users] uip200 phone not work with 1.2   Jerry Geis
• [Asterisk-Users] Call Forwarding with Account Code.. can it be done?   C F
• [Asterisk-Users] Packeteer ? Edgemark ? How to not re-cable ?   Tom Rymes
• [Asterisk-Users] Nortel Meridian Option81C to TE405P   Anish Basu
• [Asterisk-Users] Chat Lines/ Party Line Solutions for Asterisk   pdhales at optusnet.com.au
• [Asterisk-Users] Chat Lines/ Party Line Solutions for Asterisk   C F
• [Asterisk-Users] Complicated Dialing plan routing   Jan Saell
• [Asterisk-Users] Packeteer ? Edgemark ? How to not re-cable ?   Craig Guy
• [Asterisk-Users] VONAGE and Asterisk   Dakota
• [Asterisk-Users] Re: VoipBuster / Finarea   Graham Kiff
• [Asterisk-Users] Call Forwarding with Account Code.. can it be done?   Darren Wiebe
• [Asterisk-Users] DSP-based echo cancellation (T1).   Ken D'Ambrosio
• [Asterisk-Users] VONAGE and Asterisk   Erick Baum
• [Asterisk-Users] VONAGE and Asterisk   trixter aka Bret McDanel
• [Asterisk-Users] DSP-based echo cancellation (T1).   C F
• [Asterisk-Users] Flash operation on a call on a ZAP interface...   Marc-andre Poupier
• [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel   Dakota
• [Asterisk-Users] dial-out and variable inheritance problems   Philipp von Klitzing
• [Asterisk-Users] Complicated Dialing plan routing   Eric "ManxPower" Wieling
• [Asterisk-Users] can * translate DTMF from rfc2833 to inband?   Eric "ManxPower" Wieling
• [Asterisk-Users] Error when compiling asterisk   Mark Quitoriano
• [Asterisk-Users] dial-out and variable inheritance problems   Philipp von Klitzing
• [Asterisk-Users] Win up to $2000 for Asterisk Enterprise References!   jmcateer
• [Asterisk-Users] Asterisk as a Softswitch   rommel malana
• [Asterisk-Users] Asterisk as a Softswitch   Jeffery Chen
• [Asterisk-Users] Hint Priority for Polycom Phones   Douglas Garstang
• [Asterisk-Users] Fax2mail   Technical Support
• [Asterisk-Users] Connecting 2 Asterisk using SIP   Alvaro Parres
• [Asterisk-Users] Snom 360 and 320 AutoAnswer   Alvaro Parres
• [Asterisk-Users] Hint Priority for Polycom Phones   Jerry Jones
• [Asterisk-Users] Hint Priority for Polycom Phones   Adam Goryachev
• [Asterisk-Users] Hint Priority for Polycom Phones   Douglas Garstang
• [Asterisk-Users] Connecting 2 Asterisk using SIP   Waldo Rubinstein
• [Asterisk-Users] Hint Priority for Polycom Phones   Douglas Garstang
• [Asterisk-Users] Win up to $2000 for Asterisk Enterprise References!   Douglas Garstang
• [Asterisk-Users] VONAGE and Asterisk   The VoIP Connection
• [Asterisk-Users] Win up to $2000 for Asterisk Enterprise References!   BJ Weschke
• [Asterisk-Users] VONAGE and Asterisk   trixter aka Bret McDanel
• [Asterisk-Users] Win up to $2000 for Asterisk EnterpriseReferences!   Boris Bakchiev
• [Asterisk-Users] Win up to $2000 for AsteriskEnterpriseReferences!   Douglas Garstang
• [Asterisk-Users] dialplan activated Toll restriction   Craig
• [Asterisk-Users] Win up to $2000 for AsteriskEnterpriseReferences!   pdhales at optusnet.com.au
• [Asterisk-Users] Connecting asterisk over consumer wifi network   Adam Goryachev
• [Asterisk-Users] VoIPJet issue == No one is available to answer at this time   Brian Capouch
• [Asterisk-Users] audio lost on incoming calls   snacktime
• [Asterisk-Users] Snom monitoring of extensions not working   Erik
• [Asterisk-Users] Flash operation on a call on a ZAP interface...   Francesco Peeters (Asterisk)
• [Asterisk-Users] Ringtone when dialing   yusuf
• [Asterisk-Users] FAX   Bartosz Piec
• [Asterisk-Users] res_perl error when loading asterisk   Gentian Bajraktari
• [Asterisk-Users] Asterisk and Nextone MSC   dashy dude
• [Asterisk-Users] misdn, 2x HFC cards   Ivo Simicevic
• [Asterisk-Users] Feature implemention   sudhir.nayak at wipro.com
• [Asterisk-Users] Feature implemention   dashy dude
• [Asterisk-Users] diax not working properly   amna saleem
• [Asterisk-Users] Connecting 2 Asterisk using SIP   xcel
• [Asterisk-Users] Asterisk modules description   Eugene Prokopiev
• [Asterisk-Users] HDLC link unstable, yellow alarm on   Laszlo Megyer
• [Asterisk-Users] RE: OH323 user configuration   Code Lover
• [Asterisk-Users] Fax2mail   Andrew Nowrot
• [Asterisk-Users] Goldstar GDK 186 voicemail   Steve Hanselman
• [Asterisk-Users] UK ISDN2e with DDI?   Dogers
• [Asterisk-Users] Win up to $2000 for AsteriskEnterpriseReferences!   Ryan Burke
• [Asterisk-Users] Busy recognition   Ronald
• [Asterisk-Users] Win up to $2000 for Asterisk Enterprise References!   Andrew Latham
• [Asterisk-Users] HDLC link unstable, yellow alarm on   Andrew Latham
• [Asterisk-Users] VoIP to GSM?   Francesco Peeters (Asterisk)
• [Asterisk-Users] Asterisk modules description   Patrick
• [Asterisk-Users] Win up to $2000 for Asterisk Enterprise References!   Andrew Kohlsmith
• [Asterisk-Users] diax not working properly   Dan
• [Asterisk-Users] HDLC link unstable, yellow alarm on   Steve Totaro
• [Asterisk-Users] FAX   Russ Price
• [Asterisk-Users] VoIP Gateway   Adam Robins
• [Asterisk-Users] Ringtone when dialing   Steve Totaro
• [Asterisk-Users] Sending credit card thorugh network/sipura 3000   Chris Mason (Lists)
• [Asterisk-Users] Connecting asterisk over consumer wifi network   Steve Totaro
• [Asterisk-Users] asterisk with EWSD v16   Atif Rasheed
• [Asterisk-Users] Fax2mail   Technical Support
• [Asterisk-Users] FAX   Alejandro Vargas
• [Asterisk-Users] Asterisk as a Softswitch   Alejandro Vargas
• [Asterisk-Users] Connecting 2 Asterisk using SIP   Alejandro Vargas
• [Asterisk-Users] Connecting 2 Asterisk using SIP   Waldo Rubinstein
• [Asterisk-Users] Asterisk modules description   Waldo Rubinstein
• [Asterisk-Users] Busy recognition   a0305292 at unet.univie.ac.at
• [Asterisk-Users] Busy recognition   a0305292 at unet.univie.ac.at
• [Asterisk-Users] UK ISDN2e with DDI?   Patrick Lidstone (Personal E-mail)
• [Asterisk-Users] IAX or SIP   Moises Silva
• [Asterisk-Users] Complicated Dialing plan routing   Niklas Larsson
• [Asterisk-Users] VoIPJet issue == No one is available to answer at this time   David K Parker
• [Asterisk-Users] Win up to $2000 for AsteriskEnterpriseReferences!   Anish Basu
• [Asterisk-Users] Config Attended Transfer   Damian Minkov
• [Asterisk-Users] how to detect hangup   muzzamil.luqman at hauka.com
• [Asterisk-Users] RE: OH323 user configuration   Freddi Hansen
• [Asterisk-Users] E1/T1 configurations   Mark McLoughlin
• [Asterisk-Users] uip200 phone not work with 1.2   Jason Becker
• [Asterisk-Users] UK ISDN2e with DDI?   Dogers
• [Asterisk-Users] UK ISDN2e with DDI?   John Daragon
• [Asterisk-Users] AMP   Vladimir Montealegre
• [Asterisk-Users] Wanted Japan DID   Rehan Ahmed
• [Asterisk-Users] UK ISDN2e with DDI?   Dogers
• [Asterisk-Users] Polycom 501 remapping keys   a0305292 at unet.univie.ac.at
• [Asterisk-Users] Nortel Meridian Option81C to TE405P   Anthony Rodgers
• [Asterisk-Users] Connecting asterisk over consumer wifi network   Chris Bagnall
• [Asterisk-Users] Lucent TNT / Asterisk Help   Jeromy Grimmett
• [Asterisk-Users] Nortel Meridian Option81C to TE405P   Schochet, Wes
• [Asterisk-Users] IConnecthere dial out problems   John Voss
• [Asterisk-Users] UK ISDN2e with DDI?   Chris Bagnall
• [Asterisk-Users] UK ISDN2e with DDI?   John Daragon
• [Asterisk-Users] Error when compiling asterisk   jourdan lemieux
• [Asterisk-Users] Error when compiling asterisk   Kevin P. Fleming
• [Asterisk-Users] Asterisk 1.2.1 released   Remco Barende
• [Asterisk-Users] VoIP US - Toll Free Origination / Termination Providers   Waldo Rubinstein
• [Asterisk-Users] Lucent TNT / Asterisk Help   Jeromy Grimmett
• [Asterisk-Users] Asterisk 1.2.1 Released   Asterisk Development Team
• [Asterisk-Users] Zaphfc as a timing source?   Chris Bagnall
• [Asterisk-Users] Asterisk 1.2.1 released   Jared Armstrong
• [Asterisk-Users] Asterisk 1.2.1 released   Tom Vile
• [Asterisk-Users] FAX   C F
• [Asterisk-Users] FAX   C F
• [Asterisk-Users] chan_bluetooth Audio Sensitivity   Ben Higley
• [Asterisk-Users] Recording a call   Noah Silverman
• [Asterisk-Users] Nortel Meridian Option81C to TE405P   Anish Basu
• [Asterisk-Users] Can Asterisk act as a media gateway?   Ken D'Ambrosio
• [Asterisk-Users] TNT / Asterisk Help   Jeromy Grimmett
• [Asterisk-Users] SIP Video Recording   Chris
• [Asterisk-Users] Static on inside end of conversation   Jeff Busch
• [Asterisk-Users] IConnecthere dial out problems   Dennis Gilmore
• [Asterisk-Users] Static on inside end of conversation   Eric "ManxPower" Wieling
• [Asterisk-Users] Door Phones   Anton Krall
• [Asterisk-Users] Static on inside end of conversation   Steve Blair
• [Asterisk-Users] Can Asterisk act as a media gateway?   John Daragon
• [Asterisk-Users] Door Phones   C F
• [Asterisk-Users] Static on inside end of conversation   Jeff Busch
• [Asterisk-Users] Unable to compile zaptel / ztdummy   Insider KT
• [Asterisk-Users] Door Phones   Stephen Arulraj
• [Asterisk-Users] app_queue on 1.2 ?   Florian Overkamp
• [Asterisk-Users] Unable to compile zaptel / ztdummy   Tzafrir Cohen
• [Asterisk-Users] Sip behind the NAT   chawki hammoud
• [Asterisk-Users] Unable to compile zaptel / ztdummy   JP Carballo
• [Asterisk-Users] Re: Unable to compile zaptel / ztdummy   Insider KT
• [Asterisk-Users] IAX2: Don't know any of 0xf800 formats   Ryan Courtnage
• [Asterisk-Users] UK ISDN2e with DDI?   David Cook
• [Asterisk-Users] Unable to compile zaptel / ztdummy   Kunal Parikh
• [Asterisk-Users] Announcement only Devices that work with Asterisk for Dedicated 24/7 Conferencing   Kanuri, Seshu (Company IT)
• [Asterisk-Users] AstManProxy Segmentation Faults   Matt Roth
• [Asterisk-Users] UK ISDN2e with DDI?   Dogers
• [Asterisk-Users] Connecting asterisk over consumer wifi network   pdhales at optusnet.com.au
• [Asterisk-Users] UK ISDN2e with DDI?   John Daragon
• [Asterisk-Users] Recording a call   Noah Silverman
• [Asterisk-Users] Asterisk 1.2.1 and queue_log   Johann
• [Asterisk-Users] IAX2: Don't know any of 0xf800 formats   Eric "ManxPower" Wieling
• [Asterisk-Users] Help iaxmodem   Miguel Soto
• [Asterisk-Users] IAX2: Don't know any of 0xf800 formats   Ryan Courtnage
• [Asterisk-Users] IAX2: Don't know any of 0xf800 formats   Eric "ManxPower" Wieling
• [Asterisk-Users] A company that sells Toll Free Number in USA   Alvaro Parres
• [Asterisk-Users] Lucent TNT / Asterisk help   Jeromy Grimmett
• [Asterisk-Users] Recording a call   Time Bandit
• [Asterisk-Users] Announcement only Devices that work with Asterisk for Dedicated 24/7 Conferencing   C F
• [Asterisk-Users] Recording a call   Noah Silverman
• [Asterisk-Users] Recording a call   Noah Silverman
• [Asterisk-Users] Sip behind the NAT   Moises Silva
• [Asterisk-Users] Recording a call   Time Bandit
• [Asterisk-Users] app_queue on 1.2 ?   Philipp von Klitzing
• [Asterisk-Users] A company that sells Toll Free Number in USA   Rusty Dekema
• [Asterisk-Users] Recording a call   Mojo with Horan & Company, LLC
• [Asterisk-Users] Announcement only Devices that work with Asterisk for Dedicated 24/7 Conferencing   Rusty Dekema
• [Asterisk-Users] Announcement only Devices that work with Asterisk for Dedicated 24/7 Conferencing   BJ Weschke
• [Asterisk-Users] A company that sells Toll Free Number in USA   John Reynolds
• [Asterisk-Users] Recording a call   Noah Silverman
• [Asterisk-Users] Asterisk Hardware recomendation   Krystian Filiks
• [Asterisk-Users] Sip behind the NAT   Jeffery Chen
• [Asterisk-Users] Announcement only Devices that work with Asterisk for Dedicated 24/7 Conferencing   C F
• [Asterisk-Users] Re: Call Recording   Noah Silverman
• [Asterisk-Users] Asterisk Hardware recomendation   Cory Andrews
• [Asterisk-Users] Recording a call   Noah Silverman
• [Asterisk-Users] IAX2: Don't know any of 0xf800 formats   Kevin P. Fleming
• [Asterisk-Users] Recording a call   Leo Ann Boon
• [Asterisk-Users] HOW TO: CDR Customer IP address where call came in from   Rehan Ahmed
• [Asterisk-Users] RE:IConnecthere dial out problems   John Voss
• [Asterisk-Users] Aastra 9133i Configurations - are the file names to be lower case or upper case or does it matter?   Lists
• [Asterisk-Users] London DID 30 Cents a Number available - 60 Channels, ULAW   Rehan Ahmed
• [Asterisk-Users] ADIT 600 T1 with DNIS digits problem   William K. Volkman
• [Asterisk-Users] local not ring,,,,   Jeffery Chen
• [Asterisk-Users] RE:IConnecthere dial out problems   Dennis Gilmore
• [Asterisk-Users] Aastra 9133i Configurations - are the file names to be lower case or upper case or does it matter?   Robert La Ferla
• [Asterisk-Users] ZT_CHANCONFIG failed on channel 1: No such device or address   George Francis
• [Asterisk-Users] Door Phones   Héctor Garza Cantú
• [Asterisk-Users] Asterisk (OH323) - gnugk connection   Code Lover
• [Asterisk-Users] RE:how to listen voicemail messages   Tejas Shah
• [Asterisk-Users] asterisk with EWSD v16   Atif Rasheed
• [Asterisk-Users] Sip behind the NAT   chawki hammoud
• [Asterisk-Users] Recording Volume on Zap Channel   Gulzar Hussain
• [Asterisk-Users] asterisk with EWSD v16   Gulzar Hussain
• [Asterisk-Users] Asterisk as a gatekeeper   rommel malana
• [Asterisk-Users] app_queue on 1.2 ?   Florian Overkamp
• [Asterisk-Users] Asterisk Hardware recomendation   Krystian Filiks
• [Asterisk-Users] app_queue on 1.2 ?   Florian Overkamp
• [Asterisk-Users] Asterisk Hardware recomendation   Zoa
• [Asterisk-Users] HOW TO: CDR Customer IP address where call came in from   Simone Cittadini
• [Asterisk-Users] Asterisk Hardware recomendation   Krystian Filiks
• [Asterisk-Users] Asterisk Hardware recomendation   Kristian Larsson
• [Asterisk-Users] Asterisk Hardware recomendation   Roman Volf
• [Asterisk-Users] Sip behind the NAT   Jeffery Chen
• [Asterisk-Users] Echo cancellation   Kristian Larsson
• [Asterisk-Users] RE:how to listen voicemail messages   tijmen van den brink
• [Asterisk-Users] Asterisk on PPC & chan_capi issue   David Waugh
• [Asterisk-Users] Integration of external (ZAP) agents into queue   Michael Hamann
• [Asterisk-Users] Asterisk as sipclient   Morten Isaksen
• [Asterisk-Users] MYSQL cmd with Asterisk Realtime   Ah khng
• [Asterisk-Users] SVN Revision 7230   René Enskat [Teamware GmbH]
• [Asterisk-Users] A2billing signalling   Zafer Khodr
• [Asterisk-Users] dtmf problem   jibumathewemail-ast at yahoo.com
• [Asterisk-Users] Wrong caller id num on Swissvoice IP10S   Bartosz Piec
• [Asterisk-Users] FAX   Bartosz Piec
• [Asterisk-Users] asterisk with EWSD v16   Atif Rasheed
• [Asterisk-Users] CDR manipulation in macros   Herchi Silviu
• [Asterisk-Users] Streaming MOH   Stojan Sljivic - GDS
• [Asterisk-Users] GROUP_COUNT and AGI   Paradise Dove
• [Asterisk-Users] Change Inbound CALL ID "Asterisk"   Oliver Vermeulen
• [Asterisk-Users] Change Inbound CALL ID "Asterisk"   Bartosz Piec
• [Asterisk-Users] IAX stress test   Eugene Prokopiev
• [Asterisk-Users] No application 'MeetMe' for extension   Evert Meulie
• [Asterisk-Users] multiple line registrations on attendant console   Dionisis Koumouras
• [Asterisk-Users] about * and CM/CME   Andrea Riela
• [Asterisk-Users] about g729   Andrea Riela
• [Asterisk-Users] Re: Ringtone when dialing   yusuf
• [Asterisk-Users] Aastra 9133i Configurations - are the file namesto be lower case or upper case or does it matter?   lists at futuresync.com
• [Asterisk-Users] FAX   Steve Underwood
• [Asterisk-Users] Forwarding only at certain times   James Steven
• [Asterisk-Users] Forwarding only at certain times   Doug Lytle
• [Asterisk-Users] No application 'MeetMe' for extension   Kunal Parikh
• [Asterisk-Users] Re: No application 'MeetMe' for extension   Evert Meulie
• [Asterisk-Users] multiple line registrations on attendant console   Philipp von Klitzing
• [Asterisk-Users] Call simulators   Rob Hillis
• [Asterisk-Users] Asterisk Call Recording and SIP canreinvite   Waldo Rubinstein
• [Asterisk-Users] Call simulators   Simone Cittadini
• [Asterisk-Users] Asterisk Call Recording and SIP canreinvite   Steve Totaro
• [Asterisk-Users] Asterisk Call Recording and SIP canreinvite   Waldo Rubinstein
• [Asterisk-Users] Aastra 9133i Configurations - are the file namesto be lower case or upper case or does it matter?   Dave Cotton
• [Asterisk-Users] Dynamic IAX2 hosting in the UK   bails
• [Asterisk-Users] Dynamic IAX2 hosting in the UK   Simon Woodhead
• [Asterisk-Users] Echo cancellation   Steve Totaro
• [Asterisk-Users] Recording Volume on Zap Channel   Steve Totaro
• [Asterisk-Users] RE:how to listen voicemail messages   Steve Totaro
• [Asterisk-Users] Re: Odd DTMF issue over PRI   Steven
• [Asterisk-Users] A company that sells Toll Free Number in USA   Derek Whitten
• [Asterisk-Users] Hardware combination and type of asterisk configuration   David Masure
• [Asterisk-Users] Sip behind the NAT   Bharath
• [Asterisk-Users] Nortel Meridian Option81C to TE405P   Steve Totaro
• [Asterisk-Users] E1/T1 configurations   Steve Totaro
• [Asterisk-Users] Nortel Meridian Option81C to TE405P   Jerry Jones
• [Asterisk-Users] Is a BUG ? Hints and incominglimit   Paradise Dove
• [Asterisk-Users] Octo Bri card together te405p and bristuff   Kib Eki
• [Asterisk-Users] Nortel Meridian Option81C to TE405P   Joe Pukepail
• [Asterisk-Users] Polycom SIP part numbers   Alphonse Ogulla
• [Asterisk-Users] Nortel Meridian Option81C to TE405P   Steve Totaro
• [Asterisk-Users] Asterisk Call Recording and SIP canreinvite   Steve Totaro
• [Asterisk-Users] A2billing Areskicc Incoming DID setting   Sam Tam
• [Asterisk-Users] Re: Call simulators   Matt King
• [Asterisk-Users] E1/T1 configurations   Steve Underwood
• [Asterisk-Users] Nortel Meridian Option81C to TE405P   Steve Totaro
• [Asterisk-Users] Asterisk Call Recording and SIP canreinvite   Waldo Rubinstein
• [Asterisk-Users] New GSM 1-8 ports Gateway / Terminal for sale (with SMS Feature and Many more)   Sam Tam
• [Asterisk-Users] Aastra 9133i Configurations - are the file namesto be lower case or upper case or does it matter?   Robert La Ferla
• [Asterisk-Users] Exit Voicemail   Joe Pukepail
• [Asterisk-Users] New GSM 1-8 ports Gateway / Terminal for sale (with SMS Feature and Many more)   Brian Fertig
• [Asterisk-Users] Asterisk Call Recording and SIP canreinvite   Time Bandit
• [Asterisk-Users] Zombie AGI processes in FC2 / 1.2 Beta 1 under l oad   Colin Anderson
• [Asterisk-Users] Asterisk Call Recording and SIP canreinvite   Steve Totaro
• [Asterisk-Users] OOH323 towards cisco gateway (2691) call setup fails at q931: Mandatory information element is missing (96)   jacobso1
• [Asterisk-Users] Maximum Calls handled   Goran Donev
• [Asterisk-Users] New GSM 1-8 ports Gateway / Terminal for sale(with SMS Feature and Many more)   Steve Totaro
• [Asterisk-Users] Voicemail context   Benjamin Lawetz
• [Asterisk-Users] Asterisk Call Recording and SIP canreinvite   Waldo Rubinstein
• [Asterisk-Users] Exit Voicemail   C F
• [Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion   Ryan Booz
• [Asterisk-Users] Maximum Calls handled   C F
• [Asterisk-Users] Nortel Meridian Option81C to TE405P   Anish Basu
• [Asterisk-Users] Aastra 9133i Configurations - are the file names to be lower case or upper case or does it matter?   Carlos Chavez
• [Asterisk-Users] SIP.conf Technical Documentation - Help   John Voss
• [Asterisk-Users] Asterisk as a gatekeeper   Atif Rasheed
• [Asterisk-Users] Maximum Calls handled   Steve Totaro
• [Asterisk-Users] Asterisk Call Recording and SIP canreinvite   Steve Totaro
• [Asterisk-Users] Help iaxmodem   Ben Higley
• [Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion   Senad Jordanovic
• [Asterisk-Users] Realtime Replication of a Single File   Matt Roth
• [Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion   Andres
• [Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion   Dan Austin
• [Asterisk-Users] OOH323 towards cisco gateway (2691) call setupfails at q931: Mandatory information element is missing (96)   Dan Austin
• [Asterisk-Users] Asterisk Call Recording and SIP canreinvite   Noah Silverman
• [Asterisk-Users] Octo Bri card together te405p and bristuff   Tzafrir Cohen
• [Asterisk-Users] Asterisk Bounty Pool   Chris Tooley
• [Asterisk-Users] Asterisk Call Recording and SIP canreinvite   Philipp von Klitzing
• [Asterisk-Users] HDLC link unstable, yellow alarm on   janvb at caselaboratories.com
• [Asterisk-Users] Lucent MAX TNT - how do I route a DID to my sip trunk   Marc Rys
• [Asterisk-Users] SIP.conf Technical Documentation - Help   C F
• [Asterisk-Users] Realtime Replication of a Single File   burke at tailorhosting.com
• [Asterisk-Users] E1/T1 configurations   janvb at caselaboratories.com
• [Asterisk-Users] Leave one voicemail for multiple recipients   Sean Tempesta
• [Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion   Ryan Booz
• [Asterisk-Users] How do I set up extensions.conf to dial out on analog telephone line?   Robert La Ferla
• [Asterisk-Users] Re: [Asterisk-biz] Help with learning Asterisk for the real world..   JP Carballo
• [Asterisk-Users] Asterisk Bounty Pool   James Armstrong
• [Asterisk-Users] Realtime Replication of a Single File   yusuf
• [Asterisk-Users] Asterisk Bounty Pool   Chris Tooley
• [Asterisk-Users] Leave one voicemail for multiple recipients   Steve Totaro
• [Asterisk-Users] Asterisk Call Recording and SIP canreinvite   Steve Totaro
• [Asterisk-Users] How do I set up extensions.conf to dial out on analog telephone line?   burke at tailorhosting.com
• [Asterisk-Users] Asterisk Bounty Pool   Steve Totaro
• [Asterisk-Users] AstLinux 0.3.0 Released   Kristian Kielhofner
• [Asterisk-Users] Snom monitoring of extensions not working   Ivan Lopez
• [Asterisk-Users] Compile modules app_rxfax.c app_txfax.c for asterisk 1.2.1   Paul van Brouwershaven
• [Asterisk-Users] app_md5.so compile problem   René Enskat [Teamware GmbH]
• [Asterisk-Users] question about priorities?   James Armstrong
• [Asterisk-Users] Asterisk Bounty Pool   Chris Tooley
• [Asterisk-Users] Asterisk Call Recording and SIP canreinvite   Waldo Rubinstein
• [Asterisk-Users] Call screening script   Technical Support
• [Asterisk-Users] Leave one voicemail for multiple recipients   Sean Tempesta
• [Asterisk-Users] Asterisk on PPC & chan_capi issue   Patrick
• [Asterisk-Users] Re: Meetme and Sipura SPA-941 - bad jitter/distortion   Wolfgang S. Rupprecht
• [Asterisk-Users] OOH323 towards cisco gateway (2691) callsetupfails at q931: Mandatory information element is missing (96)   jacobso1
• [Asterisk-Users] OOH323 towards cisco gateway (2691)callsetupfails at q931: Mandatory information element is missing (96)   Dan Austin
• [Asterisk-Users] Exit Voicemail   Jonathan Feally
• [Asterisk-Users] Recording a call   Darrick Hartman
• [Asterisk-Users] Re: Meetme and Sipura SPA-941 - badjitter/distortion   Ryan Booz
• [Asterisk-Users] Call screening script   Innocent Evil
• [Asterisk-Users] Recording a call   C F
• [Asterisk-Users] Call screening script   C F
• [Asterisk-Users] Recording a call   Noah Silverman
• [Asterisk-Users] Nortel Meridian Option81C to TE405P   Schochet, Wes
• [Asterisk-Users] Where to find Digium products in Mexico?   Claudio Canseco
• [Asterisk-Users] Realtime Replication of a Single File   JP Carballo
• [Asterisk-Users] Aastra 9133i Configurations - are the filenames to be lower case or upper case or does it matter?   Lists
• [Asterisk-Users] SRV lookups   Douglas Garstang
• [Asterisk-Users] Recording a call   Darrick Hartman
• [Asterisk-Users] OOH323 towards cisco gateway(2691)callsetupfails at q931: Mandatory information element ismissing (96)   jacobso1
• [Asterisk-Users] OOH323 towards cisco gateway(2691)callsetupfailsat q931: Mandatory information element ismissing (96)   Dan Austin
• [Asterisk-Users] Recording a call   C F
• [Asterisk-Users] dial problem   Peter Loeppky
• [Asterisk-Users] Re: Meetme and Sipura SPA-941 -badjitter/distortion   Dan Austin
• [Asterisk-Users] "Unable to transfer"   Miguel Soto
• [Asterisk-Users] Asterisk and Adtran TA 750 Channel Bank -- odd behavior (help!)   Gaurav Naik
• [Asterisk-Users] [MSG]TDM Error on ASUS Pundit-R   amx109
• [Asterisk-Users] Why Won't Asterisk REINVITE?   George Pajari
• [Asterisk-Users] Exit Voicemail   KRTorio
• [Asterisk-Users] Why Won't Asterisk REINVITE?   C F
• [Asterisk-Users] OOH323 towards cisco gateway(2691)callsetupfailsat q931: Mandatory information element ismissing (96)   Jeffery Chen
• [Asterisk-Users] ztdummy on FC4   Jim Duda
• [Asterisk-Users] cant start the conference bride   chan
• [Asterisk-Users] cant start the conference bride   pdhales at optusnet.com.au
• [Asterisk-Users] Where to find Digium products in Mexico?   Moises Silva
• [Asterisk-Users] Asterisk Bounty Pool   Chris Tooley
• [Asterisk-Users] does asterisk-oh323-0.6.7support asterisk1.2   =?gb2312?B?cGFvcGFvZXJ6aGFuZyiPiNa+w/c90dCwbCk=?=
• [Asterisk-Users] Core dumps since 1.2.0   Ryan Laginski
• [Asterisk-Users] does asterisk-oh323-0.6.7support asterisk1.2   Jeffery Chen
• [Asterisk-Users] OOH323 towards ciscogateway(2691)callsetupfailsat q931: Mandatory informationelement ismissing (96)   jacobso1
• [Asterisk-Users] Why Won't Asterisk REINVITE?   Eric "ManxPower" Wieling
• [Asterisk-Users] SRV lookups   Eric "ManxPower" Wieling
• [Asterisk-Users] Asterisk Dial Failover   Douglas Garstang
• [Asterisk-Users] Why Won't Asterisk REINVITE?   George Pajari
• [Asterisk-Users] Can Asterisk accept and relay calls   Skeeve Stevens
• [Asterisk-Users] Can Asterisk accept and relay calls   Kerry Garrison
• [Asterisk-Users] Porting a phone number to a voip provider   Matthew
• [Asterisk-Users] Can Asterisk accept and relay calls   pdhales at optusnet.com.au
• [Asterisk-Users] DID Providers   Aaron Anderson
• [Asterisk-Users] Asterisk on PPC & chan_capi issue   David Waugh
• [Asterisk-Users] PRI billing signalization   Tomislav Parčina
• [Asterisk-Users] Re: Is Polycom 500CS with P# 2201.11500.001 SIP capable?   Alphonse Ogulla
• [Asterisk-Users] Re: No application 'MeetMe' for extension   Evert Meulie
• [Asterisk-Users] Re: ztdummy on FC4   Tony Mountifield
• [Asterisk-Users] Re: Core dumps since 1.2.0   Tony Mountifield
• [Asterisk-Users] Sip behind the NAT   Wilson Pickett
• [Asterisk-Users] SIP Canreinvite   Giordano Grandis
• [Asterisk-Users] Aastra firmware 1.3.x. Solution to Far-End sound level issue   BennyBad
• [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel   asterisk-users at ecessor.com
• [Asterisk-Users] Call simulators   Lenz
• [Asterisk-Users] /dev/zap/ctl or /dev/zapctl cause ztdummy in init.d failed   L
• [Asterisk-Users] Bristuff / Junghanns / Customer Service   Frederic Steinfels
• [Asterisk-Users] Hangup after dialing   René Enskat [Teamware GmbH]
• [Asterisk-Users] Queue routing - calls return to agent which previously handled call   Hilton Williams
• [Asterisk-Users] CIDNUM CIDNAME   René Enskat [Teamware GmbH]
• [Asterisk-Users] Re: /dev/zap/ctl or /dev/zapctl cause ztdummy in init.d failed   Tony Mountifield
• [Asterisk-Users] Re: /dev/zap/ctl or /dev/zapctl cause ztdummy in init.d failed - SOLVED   L
• [Asterisk-Users] /dev/zap/ctl or /dev/zapctl cause ztdummy in init.d failed   Steve Ringwald
• [Asterisk-Users] Asterisk Dial Failover   Philipp von Klitzing
• [Asterisk-Users] connection between asterisk and cisco   muhammad usman
• [Asterisk-Users] Asterisk on PPC & chan_capi issue   Armin Schindler
• [Asterisk-Users] Asterisk Dial Failover   burke at tailorhosting.com
• [Asterisk-Users] Porting a phone number to a voip provider   Robert Webb
• [Asterisk-Users] Re: ztdummy on FC4   Jim Duda
• [Asterisk-Users] Queue routing - calls return to agent which previously handled call   Lenz
• [Asterisk-Users] Porting a phone number to a voip provider   Rich Adamson
• [Asterisk-Users] Asterisk Dial Failover   Jonathan k. Creasy
• [Asterisk-Users] Change time when * is running   Julian Lyndon-Smith
• [Asterisk-Users] Porting a phone number to a voip provider   Matthew
• [Asterisk-Users] Polycom 501 remapping keys   Matthew
• [Asterisk-Users] Queue routing - calls return to agent which previously handled call   Rob Lith
• [Asterisk-Users] Change time when * is running   Kevin P. Fleming
• [Asterisk-Users] Low Layer Compatibility (LLC) not forwarded?   Lars Poschitzki
• [Asterisk-Users] Wait for X rings before answering?   José Luis Gómez
• [Asterisk-Users] Change time when * is running   Tzafrir Cohen
• [Asterisk-Users] Change time when * is running   Gavin Hamill
• [Asterisk-Users] Low Layer Compatibility (LLC) not forwarded?   Klaus Darilion
• [Asterisk-Users] Porting a phone number to a voip provider   Rich Adamson
• [Asterisk-Users] Change time when * is running   Julian Lyndon-Smith
• [Asterisk-Users] Change time when * is running   Julian Lyndon-Smith
• [Asterisk-Users] Change time when * is running   Gavin Hamill
• [Asterisk-Users] Re: ztdummy on FC4   Tony Mountifield
• [Asterisk-Users] Why Won't Asterisk REINVITE?   Julian J. M.
• [Asterisk-Users] Asterisk Dial Failover   Adam Robins
• [Asterisk-Users] Phone Information   James Horn
• [Asterisk-Users] New GSM 1-8 ports Gateway / Terminal for sale (with SMS Feature and Many more)   Rusty Dekema
• [Asterisk-Users] Asterisk Dial Failover   John Cianfarani
• [Asterisk-Users] Asterisk Dial Failover   Douglas Garstang
• [Asterisk-Users] Wait for X rings before answering?   Robert La Ferla
• [Asterisk-Users] Teliax experiences   Rolf Brusletto
• [Asterisk-Users] Asterisk Dial Failover   Douglas Garstang
• [Asterisk-Users] X100 clone   Vladimir Montealegre
• [Asterisk-Users] Porting a phone number to a voip provider   Rich Adamson
• [Asterisk-Users] Echo PSTN Asterisk@Home 2.0 Digium TDM11B & DSL   David K Parker
• [Asterisk-Users] Asterisk on PPC & chan_capi issue   Jason Williams
• [Asterisk-Users] Teliax experiences   Rich Adamson
• [Asterisk-Users] Asterisk Dial Failover   Adam Robins
• [Asterisk-Users] Asterisk Dial Failover   Ashley Wright
• [Asterisk-Users] Asterisk Dial Failover   Douglas Garstang
• [Asterisk-Users] Re: ztdummy on FC4   Jim Duda
• [Asterisk-Users] Phone Information   C F
• [Asterisk-Users] Asteriskguru Queue Statistics version 0.7 released   Zoa
• [Asterisk-Users] Re: ztdummy on FC4   Jim Duda
• [Asterisk-Users] CIDNUM CIDNAME   C F
• [Asterisk-Users] Asterisk Dial Failover   burke at tailorhosting.com
• [Asterisk-Users] Wait for X rings before answering?   Derek Whitten
• [Asterisk-Users] a few questions   Stas Khromoy
• [Asterisk-Users] Teliax experiences   John Reynolds
• [Asterisk-Users] Change time when * is running   Remco Barende
• [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel   O'Connor, Jonathan
• [Asterisk-Users] a few questions   Kerry Garrison
• [Asterisk-Users] Wait for X rings before answering?   Robert La Ferla
• [Asterisk-Users] a few questions   Kerry Garrison
• [Asterisk-Users] Wait for X rings before answering?   Kerry Garrison
• [Asterisk-Users] Phone Information   Time Bandit
• [Asterisk-Users] Phone Information   Lawrence Jovellanos
• [Asterisk-Users] t38 support in latest asterisk release   Erick Perez
• [Asterisk-Users] Re: ztdummy on FC4   Tony Mountifield
• [Asterisk-Users] Wait for X rings before answering?   Dave Cotton
• [Asterisk-Users] testers needed for channel.c jitter buffer (better known as SIP jitter buffer)   Zoa
• [Asterisk-Users] DID Providers   Rehan Ahmed
• [Asterisk-Users] Nortel Meridian Option81C to TE405P   Schochet, Wes
• [Asterisk-Users] Phone Information   Jeffery Chen
• [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel   Schochet, Wes
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 56   James Horn
• [Asterisk-Users] Teliax experiences   Chris
• [Asterisk-Users] a few questions   C F
• [Asterisk-Users] a few questions   C F
• [Asterisk-Users] Porting a phone number to a voip provider   Austin Denyer
• [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel   Colin Anderson
• [Asterisk-Users] Porting a phone number to a voip provider   Brian Capouch
• [Asterisk-Users] Wait for X rings before answering?   Robert La Ferla
• [Asterisk-Users] Asterisk and Adtran TA 750 Channel Bank -- odd behavior (help!)   Gaurav Naik
• [Asterisk-Users] Re: ztdummy on FC4   Jim Duda
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 56   C F
• [Asterisk-Users] a few questions   Jerry Jones
• [Asterisk-Users] Wait for X rings before answering?   Derek Whitten
• [Asterisk-Users] Wait for X rings before answering?   Derek Whitten
• [Asterisk-Users] Hardware based echo can from NMS Communications   C F
• [Asterisk-Users] Asterisk Dial Failover   Michiel van Baak
• [Asterisk-Users] Wait for X rings before answering?   burke at tailorhosting.com
• [Asterisk-Users] Asterisk Bounty Pool   Rehan Ahmed
• [Asterisk-Users] a few questions   Kerry Garrison
• [Asterisk-Users] Asterisk Bounty Pool   Chris Tooley
• [Asterisk-Users] Re: ztdummy on FC4   Tony Mountifield
• [Asterisk-Users] Wait for X rings before answering?   Dave Cotton
• [Asterisk-Users] Re: Meetme and Sipura SPA-941-badjitter/distortion   Ryan Booz
• [Asterisk-Users] Wait for X rings before answering?   C Stephens
• [Asterisk-Users] Teliax experiences   David K Parker
• [Asterisk-Users] a few questions   Dean Collins
• [Asterisk-Users] Wait for X rings before answering?   Time Bandit
• [Asterisk-Users] Unknown RTP codec 96 received   Javier Oviedo
• [Asterisk-Users] Teliax experiences   Darrick Hartman
• [Asterisk-Users] T.38/PRI Gateway Recommendations   George Pajari
• [Asterisk-Users] does asterisk-oh323-0.6.7support asterisk1.2   Tzafrir Cohen
• [Asterisk-Users] IAX2 Status monitoring   Rolf Brusletto
• [Asterisk-Users] X100 clone   Tzafrir Cohen
• [Asterisk-Users] IAX2 Status monitoring   Juan Janczuk
• [Asterisk-Users] Asterisk 1.2 chan_misdn modules.conf   Kristian Kielhofner
• [Asterisk-Users] number of users in a meetme conference   imran ahmed
• [Asterisk-Users] IAX2 Status monitoring   Rolf Brusletto
• [Asterisk-Users] Asterisk 1.2 chan_misdn modules.conf   Joshua Colp
• [Asterisk-Users] IAX2 Status monitoring   Juan Janczuk
• [Asterisk-Users] Asterisk 1.2 chan_misdn modules.conf   Kristian Kielhofner
• [Asterisk-Users] Sip behind the NAT   Tom Rymes
• [Asterisk-Users] Adit 600 and Centrex   Doug Lytle
• [Asterisk-Users] Change time when * is running   Tzafrir Cohen
• [Asterisk-Users] a few questions   Stas Khromoy
• [Asterisk-Users] Call Routing from GnuGK to Asterisk   Code Lover
• [Asterisk-Users] legacy pbx   James Sturges
• [Asterisk-Users] Asterisk Consulant - Call Centre Proposal   James Sturges
• [Asterisk-Users] Wait for X rings before answering?   Eric "ManxPower" Wieling
• [Asterisk-Users] Wait for X rings before answering?   Eric "ManxPower" Wieling
• [Asterisk-Users] Adit 600 and Centrex   Rich Adamson
• [Asterisk-Users] Wait for X rings before answering?   Eric "ManxPower" Wieling
• [Asterisk-Users] callerid international-format   Florian Meister
• [Asterisk-Users] Asterisk Consulant - Call Centre Proposal   pdhales at optusnet.com.au
• [Asterisk-Users] Wait for X rings before answering?   pdhales at optusnet.com.au
• [Asterisk-Users] Adit 600 and Centrex   Doug Lytle
• [Asterisk-Users] a few questions   pdhales at optusnet.com.au
• [Asterisk-Users] Adit 600 and Centrex   Rich Adamson
• [Asterisk-Users] Wait for X rings before answering?   Rich Adamson
• [Asterisk-Users] NY Clec Listing   Rehan Ahmed
• [Asterisk-Users] Stop Dial but no Hangup   Benoît Mérouze
• [Asterisk-Users] Unable to receive DTMF inbound   Don Fanning
• [Asterisk-Users] T.38/PRI Gateway Recommendations   Peder at NetworkOblivion
• [Asterisk-Users] Echo PSTN Asterisk@Home 2.0 Digium TDM11B & DSL   Matthew Fredrickson
• [Asterisk-Users] IAX Jitterbuffer and trunking   Richard Scobie
• [Asterisk-Users] Anyone else experiencing Voicepulse outage?   Ben Higley
• [Asterisk-Users] HOW TO: CDR Customer IP address where call came in from   Rehan Ahmed
• [Asterisk-Users] Adit 600 and Centrex   Doug Lytle
• [Asterisk-Users] Daily Reboot Script for Asterisk Question   Min Hwan Chang
• [Asterisk-Users] Daily Reboot Script for Asterisk Question   Pete Barnwell
• [Asterisk-Users] IAX Jitterbuffer and trunking   Philipp von Klitzing
• [Asterisk-Users] CDR problem   Edwin Lam
• [Asterisk-Users] Asterisk, Small Business, and Teliax   Andrew Berman
• [Asterisk-Users] chan_bluetooth and Ericsson/SonyEricsson models   David Woodhouse
• [Asterisk-Users] Synthesized Voice for Asterisk   Dakota
• [Asterisk-Users] Major Advantages of Asterisk vs Nortel PBX Systems   Dakota
• [Asterisk-Users] Asterisk, Small Business, and Teliax   Manny A. Wise
• [Asterisk-Users] Synthesized Voice for Asterisk   Robert La Ferla
• [Asterisk-Users] Synthesized Voice for Asterisk   snacktime
• [Asterisk-Users] Daily Reboot Script for Asterisk Question   Mojo with Horan & Company, LLC
• [Asterisk-Users] Synthesized Voice for Asterisk   Robert La Ferla
• [Asterisk-Users] sellvoip.net service   Tom Vile
• [Asterisk-Users] Major Advantages of Asterisk vs Nortel PBX Systems   Dean Collins
• [Asterisk-Users] Major Advantages of Asterisk vs Nortel PBX Systems   Arnaud
• [Asterisk-Users] Asterisk, Small Business, and Teliax   John Reynolds
• [Asterisk-Users] Synthesized Voice for Asterisk   John Cianfarani
• [Asterisk-Users] Asterisk without RTP streams vs. SER in statefull mode   ihatewindowsxp at gmail.com
• [Asterisk-Users] What's the best opensource web interface for customer portal   ihatewindowsxp at gmail.com
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 56   pdhales at optusnet.com.au
• [Asterisk-Users] Dial Command Doesn't return Correctly! (Bug?)   Douglas Garstang
• [Asterisk-Users] RE: Dial Command Doesn't return Correctly! (Bug?)   Douglas Garstang
• [Asterisk-Users] Asterisk, Small Business, and Teliax   Kerry Garrison
• [Asterisk-Users] Daily Reboot Script for Asterisk Question   Roger Hill
• [Asterisk-Users] callerid international-format   Florian Overkamp
• [Asterisk-Users] Via Epia   Andrew Nowrot
• [Asterisk-Users] Predefined Channel Variables   Code Lover
• [Asterisk-Users] Q regarding dialling multiple phones (fwd)   Bob Purdon
• [Asterisk-Users] Via Epia   Maciej Kietlinski
• [Asterisk-Users] Sending a recorded message to voicemail   Steve Hanselman
• [Asterisk-Users] Via Epia   Roger Hill
• [Asterisk-Users] Asterisk, Small Business, and Teliax   Rich Adamson
• [Asterisk-Users] Channel 0/1, span 1 got hangup request   Anton Bakulev
• [Asterisk-Users] Change time when * is running   Alejandro Vargas
• [Asterisk-Users] Email to voice?   Chris Mason (Lists)
• [Asterisk-Users] Possible bug in record?   Steve Hanselman
• [Asterisk-Users] Asterisk not Replying on Port Specified in the VIA header   Saad Siddiqi
• [Asterisk-Users] Email to voice?   Rich Adamson
• [Asterisk-Users] Possible bug in record?   Doug Lytle
• [Asterisk-Users] Channel 0/1, span 1 got hangup request   Steve Totaro
• [Asterisk-Users] Via Epia   Roger Hill
• [Asterisk-Users] Daily Reboot Script for Asterisk Question   Doug Lytle
• [Asterisk-Users] Synthesized Voice for Asterisk   Steve Totaro
• [Asterisk-Users] Via Epia   Sergey Okhapkin
• [Asterisk-Users] Via Epia   Sergey Okhapkin
• [Asterisk-Users] Email to voice?   Jean-Michel Hiver
• [Asterisk-Users] Email to voice?   Peter Bowyer
• [Asterisk-Users] Asterisk, Small Business, and Teliax   Andrew Berman
• [Asterisk-Users] Call Routing from GnuGK to Asterisk   Rafael Marconi
• [Asterisk-Users] Via Epia   Andrew Nowrot
• [Asterisk-Users] Asterisk, Small Business, and Teliax   Rich Adamson
• [Asterisk-Users] Asterisk Bounty Pool   Chris Tooley
• [Asterisk-Users] Door Phones   Anton Krall
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61   Abhishek Gangal
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61   Steve Totaro
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61   Steve Underwood
• [Asterisk-Users] agi variables list   Olivier Taylor
• [Asterisk-Users] Problems with zaptel channels not properly being answered...   Jim Hodapp
• [Asterisk-Users] agi variables list   Steve Totaro
• [Asterisk-Users] Problems with zaptel channels not properly beinganswered...   Steve Totaro
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61   Ossi Sariola
• [Asterisk-Users] agi variables list   Olivier Taylor
• [Asterisk-Users] Asterisk, Small Business, and Teliax   Andrew Berman
• [Asterisk-Users] Good Dialing Macros   gw at adcomcorp.com
• [Asterisk-Users] Problems with zaptel channels not properly being answered...   Rich Adamson
• [Asterisk-Users] agi variables list   Asterisk [Submusic]
• [Asterisk-Users] Call Routing from GnuGK to Asterisk   Code Lover
• [Asterisk-Users] Via Epia   Tzafrir Cohen
• [Asterisk-Users] Re: Teliax experiences   Asterisk Users - Dovid
• [Asterisk-Users] What's the best opensource web interface for customer portal   Dan Littlejohn
• [Asterisk-Users] extensions and regular expressions ( probably an easy question )   Sean Kennedy
• [Asterisk-Users] Echo on incoming sip provider- asterisk - sip phone / ata   Bernard
• [Asterisk-Users] What's the best opensource web interface for customer portal   Tom Vile
• [Asterisk-Users] RE: chan_grab.c for version 1.0X   Ronald Hartmann
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61   Brian Capouch
• [Asterisk-Users] Via Epia   Andrew Nowrot
• [Asterisk-Users] Using SIP_HEADER() Function correctly   Rushowr
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61   pdhales at optusnet.com.au
• [Asterisk-Users] Updated Guide to SPA-3000   Kerry Garrison
• [Asterisk-Users] extensions and regular expressions ( probably an easy question )   Daniel Wright
• [Asterisk-Users] mpg123 on x86_64 (Opteron MP)   Brad
• [Asterisk-Users] extensions and regular expressions ( probably an easy question )   Sean Kennedy
• [Asterisk-Users] extensions and regular expressions ( probablyan easy question )   Steve Totaro
• [Asterisk-Users] extensions and regular expressions ( probably an easy question )   Daniel Wright
• [Asterisk-Users] extensions and regular expressions ( probablyan easy question )   Rushowr
• [Asterisk-Users] extensions and regular expressions ( probably an easy question )   Rich Adamson
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 56   C F
• [Asterisk-Users] extensions and regular expressions ( probablyan easy question )   Sean Kennedy
• [Asterisk-Users] a few questions   C F
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 56   pdhales at optusnet.com.au
• [Asterisk-Users] Setting Request URI   Douglas Garstang
• [Asterisk-Users] Setting Request URI   Marc Storck
• [Asterisk-Users] Echo PSTN Asterisk@Home 2.0 Digium TDM11B & DSL   David K Parker
• [Asterisk-Users] extensions and regular expressions ( probablyan easy question )   Ryan Burke
• [Asterisk-Users] extensions and regular expressions ( probably an easy question )   Sean Kennedy
• [Asterisk-Users] MeetMe questions   Schochet, Wes
• [Asterisk-Users] bristuff use without BRI/PRI   Robert Lawrence
• [Asterisk-Users] Setting Request URI   Douglas Garstang
• [Asterisk-Users] Via Epia   Tzafrir Cohen
• [Asterisk-Users] Email to voice?   Chris Mason (Lists)
• [Asterisk-Users] Email to voice?   Chris Mason (Lists)
• [Asterisk-Users] mpg123 on x86_64 (Opteron MP)   Tzafrir Cohen
• [Asterisk-Users] Email to voice?   Peter Bowyer
• [Asterisk-Users] Channel 0/1, span 1 got hangup request   Антон Бакулев
• [Asterisk-Users] bristuff use without BRI/PRI   Harald Holzer
• [Asterisk-Users] FAX   Mark Quitoriano
• [Asterisk-Users] Asterisk not Replying on Port Specified in the VIA header   Saad Siddiqi
• [Asterisk-Users] extensions and regular expressions ( probably an easy question )   Rich Adamson
• [Asterisk-Users] MeetMe questions   Rich Adamson
• [Asterisk-Users] FAX   Steve Underwood
• [Asterisk-Users] MeetMe questions   Doug Lytle
• [Asterisk-Users] outgoing calls that last an unreasonably long time   Warren Burstein
• [Asterisk-Users] bristuff use without BRI/PRI   Chris Bagnall
• [Asterisk-Users] bristuff use without BRI/PRI   Michiel van Baak
• [Asterisk-Users] Dialing analog extensions from SIP?   Robert La Ferla
• [Asterisk-Users] SRV Lookups *ARRGH!*   Douglas Garstang
• [Asterisk-Users] Dialing analog extensions from SIP?   Doug Lytle
• [Asterisk-Users] outgoing calls that last an unreasonably long time   Simone Cittadini
• [Asterisk-Users] Dialing analog extensions from SIP?   Robert La Ferla
• [Asterisk-Users] Dialing analog extensions from SIP?   Eric "ManxPower" Wieling
• [Asterisk-Users] Dialing analog extensions from SIP?   Robert La Ferla
• [Asterisk-Users] Dialing analog extensions from SIP?   Doug Lytle
• [Asterisk-Users] SRV Lookups *ARRGH!*   Philipp von Klitzing
• [Asterisk-Users] SRV Lookups *ARRGH!*   Douglas Garstang
• [Asterisk-Users] outgoing calls that last an unreasonably long time   Warren Burstein
• [Asterisk-Users] SRV Lookups   Douglas Garstang
• [Asterisk-Users] SIP Qualify   Douglas Garstang
• [Asterisk-Users] Asterisk Dynamic DNS   Branko Samardzic
• [Asterisk-Users] [Asterisk-Dev] C++ AGI debuggin   Danish Samad
• [Asterisk-Users] Asterisk Dynamic DNS   Tom Vile
• [Asterisk-Users] Asterisk Dynamic DNS   Roger Hill
• [Asterisk-Users] Asterisk Dynamic DNS   Schochet, Wes
• [Asterisk-Users] Asterisk Dynamic DNS   Sergio Chersovani
• [Asterisk-Users] SRV Lookups   Leif Madsen
• [Asterisk-Users] Channel 0/1, span 1 got hangup request   Steve Totaro
• [Asterisk-Users] Asterisk Dynamic DNS   Branko Samardzic
• [Asterisk-Users] Attack dialing   Eric Bishop
• [Asterisk-Users] SRV Lookups   Douglas Garstang
• [Asterisk-Users] SRV Lookups   Douglas Garstang
• [Asterisk-Users] Attack dialing   Gavin Hamill
• [Asterisk-Users] Re: Teliax experiences   gw at adcomcorp.com
• [Asterisk-Users] SRV Lookups *ARRGH!*   Matt Riddell
• [Asterisk-Users] Attack dialing   Michael Welter
• [Asterisk-Users] SRV Lookups   Kevin P. Fleming
• [Asterisk-Users] SRV Lookups   Paul Tinsley
• [Asterisk-Users] SRV Lookups   Douglas Garstang
• [Asterisk-Users] Attack dialing   Gary Reuter
• [Asterisk-Users] SRV Lookups   Aaron Daniel
• [Asterisk-Users] mpg123 on x86_64 (Opteron MP)   Brad
• [Asterisk-Users] Change time when * is running   Julian Lyndon-Smith
• [Asterisk-Users] Regexten   Douglas Garstang
• [Asterisk-Users] SRV Lookups   Kerry Garrison
• [Asterisk-Users] SRV Lookups   Douglas Garstang
• [Asterisk-Users] Regexten   Joshua Colp
• [Asterisk-Users] SRV Lookups   Douglas Garstang
• [Asterisk-Users] Regexten   Douglas Garstang
• [Asterisk-Users] Attack dialing   C F
• [Asterisk-Users] Asterisk Dynamic DNS   Wilson Pickett
• [Asterisk-Users] SRV Lookups   Kevin P. Fleming
• [Asterisk-Users] Attack dialing   pdhales at optusnet.com.au
• [Asterisk-Users] Asterisk Dynamic DNS   Branko Samardzic
• [Asterisk-Users] SRV Lookups   Eric "ManxPower" Wieling
• [Asterisk-Users] SRV Lookups *ARRGH!*   Philipp von Klitzing
• [Asterisk-Users] mpg123 on x86_64 (Opteron MP)   snacktime
• [Asterisk-Users] Asterisk Limitations   Eric "ManxPower" Wieling
• [Asterisk-Users] Realtime Subscribecontext   Douglas Garstang
• [Asterisk-Users] Small / embedded system recommendations   Alistair Cunningham
• [Asterisk-Users] Attack dialing   Rich Adamson
• [Asterisk-Users] Realtime Subscribecontext   Kevin P. Fleming
• [Asterisk-Users] Realtime Subscribecontext   Douglas Garstang
• [Asterisk-Users] Realtime Subscribecontext   Kevin P. Fleming
• [Asterisk-Users] Attack dialing   trixter aka Bret McDanel
• [Asterisk-Users] Attack dialing   Alistair Cunningham
• [Asterisk-Users] Asterisk Limitations   James Sturges
• [Asterisk-Users] Asterisk Limitations   trixter aka Bret McDanel
• [Asterisk-Users] Regexten   Leif Madsen
• [Asterisk-Users] Asterisk Dynamic DNS   John Biundo
• [Asterisk-Users] Reboot stops TD400P cards from outgoing calls until first incoming call arrives - resolution   James B. MacLean
• [Asterisk-Users] Re: Email to voice?   Steven
• [Asterisk-Users] Small / embedded system recommendations   Chris Bagnall
• [Asterisk-Users] Cleaning up CLID on incoming PRI lines   Adrian Carter
• [Asterisk-Users] Re: Email to voice?   Jeffery Chen
• [Asterisk-Users] Originate action script causes Asterisk to hang   KRTorio
• [Asterisk-Users] Small / embedded system recommendations   Kristian Kielhofner
• [Asterisk-Users] Cleaning up CLID on incoming PRI lines   Kristian Kielhofner
• [Asterisk-Users] Asterisk Dynamic DNS   Branko Samardzic
• [Asterisk-Users] Regexten   Douglas Garstang
• [Asterisk-Users] Realtime Subscribecontext   Douglas Garstang
• [Asterisk-Users] SRV Lookups   Douglas Garstang
• [Asterisk-Users] Realtime Subscribecontext   Douglas Garstang
• [Asterisk-Users] extensions and regular expressions ( probablyan easy question )   Rushowr
• [Asterisk-Users] SRV Lookups   Kerry Garrison
• [Asterisk-Users] Mechanisms for Implementing a Common Contact Database   Douglas Garstang
• [Asterisk-Users] New Product ID.   Goran Donev
• [Asterisk-Users] Major Advantages of Asterisk vs Nortel PBX Systems   Rushowr
• [Asterisk-Users] SRV Lookups   Douglas Garstang
• [Asterisk-Users] Mechanisms for Implementing a Common ContactDatabase   Douglas Garstang
• [Asterisk-Users] zapata directory not found in svn .   f6hqz-m at hamwlan.net
• [Asterisk-Users] Long and variable echo   James Andrewartha
• [Asterisk-Users] CallerID Transfer   asterisk183
• [Asterisk-Users] CallerID Transfer   Rushowr
• [Asterisk-Users] asterisk1.2.1+realtimedb+voicemail+contexts   Frank Aartman
• [Asterisk-Users] [helpp] Problem in astersik   Talat Ishtiaq
• [Asterisk-Users] Long and variable echo   Kristof Hardy
• [Asterisk-Users] Problems with current chan-capi-cm   Peer Oliver Schmidt
• [Asterisk-Users] "Got clone lock for masquerade" crash   Benny Amorsen
• [Asterisk-Users] [helpp] Problem in astersik   Roger Hill
• [Asterisk-Users] Cisco 7941 difference   Tomislav Parčina
• [Asterisk-Users] Asterisk cluster and astdb   Tomislav Parcina
• [Asterisk-Users] Variables naming, may be a BUG??   Marcello Lupo
• [Asterisk-Users] click to dial applications   Joseph Rothstein
• [Asterisk-Users] Problem with Speex   hrishikesh shrivastaw
• [Asterisk-Users] [helpp] Problem in astersik   Mark Edwards
• [Asterisk-Users] PRI E1 - HDLC Bad FCS / HDLC Abort errors   Dushyanth Harinath
• [Asterisk-Users] [helpp] Problem in astersik   Talat Ishtiaq
• [Asterisk-Users] Re: "Got clone lock for masquerade" crash   Benny Amorsen
• [Asterisk-Users] [helpp] Problem in astersik   asterisk
• [Asterisk-Users] Problems with current chan-capi-cm   Armin Schindler
• [Asterisk-Users] Long and variable echo   pdhales at optusnet.com.au
• [Asterisk-Users] [helpp] Problem in astersik   Talat Ishtiaq
• [Asterisk-Users] Problems with current chan-capi-cm   Peer Oliver Schmidt
• [Asterisk-Users] [helpp] Problem in astersik   Kiss Karoly
• [Asterisk-Users] Re: Problem with Speex   Steven
• [Asterisk-Users] Re: click to dial applications   Steven
• [Asterisk-Users] Re: [helpp] Problem in astersik   Steven
• [Asterisk-Users] Mechanisms for Implementing a Common Contact Database   Philipp von Klitzing
• [Asterisk-Users] Production Upgrades   Steven
• [Asterisk-Users] [helpp] Problem in astersik   Talat Ishtiaq
• [Asterisk-Users] Digium PCI-X timeline   Steven
• [Asterisk-Users] Asterisk Dial Failover   Patrick
• [Asterisk-Users] Asterisk on PPC & chan_capi issue   Patrick
• [Asterisk-Users] Asterisk Dynamic DNS   Matt Riddell
• [Asterisk-Users] Re: Asterisk Users Newsgroup   Tomislav Parcina
• [Asterisk-Users] Problems with current chan-capi-cm   Armin Schindler
• [Asterisk-Users] mpg123 on x86_64 (Opteron MP)   Patrick
• [Asterisk-Users] Problems with current chan-capi-cm   Peer Oliver Schmidt
• [Asterisk-Users] Small / embedded system recommendations   Alistair Cunningham
• [Asterisk-Users] Small / embedded system recommendations   Alistair Cunningham
• [Asterisk-Users] Small / embedded system recommendations   BJ Weschke
• [Asterisk-Users] Long and variable echo   Eric "ManxPower" Wieling
• [Asterisk-Users] persistentagents, persistentmembers   Dov Bigio
• [Asterisk-Users] Attack dialing   Matt
• [Asterisk-Users] Synthesized Voice for Asterisk   Matt
• [Asterisk-Users] ChefSec function   René Enskat [Teamware GmbH]
• [Asterisk-Users] RE: CallerID Transfer   Tomislav Parcina
• [Asterisk-Users] Zap Transfer   Sean Cook
• [Asterisk-Users] Zap Transfer   Sean Cook
• [Asterisk-Users] ASTCC/ASTCC anything wrong with that?   bbench at mail.bg
• [Asterisk-Users] Attack dialing   trixter aka Bret McDanel
• [Asterisk-Users] Realtime Subscribecontext   Kevin P. Fleming
• [Asterisk-Users] ASTCC/ASTCC anything wrong with that?   Darren Wiebe
• [Asterisk-Users] Mechanisms for Implementing a Common ContactDatabase   Kevin P. Fleming
• [Asterisk-Users] zapata directory not found in svn .   Kevin P. Fleming
• [Asterisk-Users] How do I remove the temp greeting?!?!   Matt
• [Asterisk-Users] Digium PCI-X timeline   Kevin P. Fleming
• [Asterisk-Users] How do I remove the temp greeting?!?!   Steve Blair
• [Asterisk-Users] How do I remove the temp greeting?!?!   James Armstrong
• [Asterisk-Users] Digium PCI-X timeline   Steve Underwood
• [Asterisk-Users] Realtime Subscribecontext   Denis Galvão - iSolve
• [Asterisk-Users] RE:IConnecthere dial out problems   John Voss
• [Asterisk-Users] How do I remove the temp greeting?!?!   Matt
• [Asterisk-Users] Attack dialing   Matt
• [Asterisk-Users] Synthesized Voice for Asterisk   Steve Totaro
• [Asterisk-Users] mpg123 on x86_64 (Opteron MP)   Steve Totaro
• [Asterisk-Users] mpg123 on x86_64 (Opteron MP)   Jason Becker
• [Asterisk-Users] Dial Cmd Outbound CLID Failure (* 1.2.1)   Darren Sessions
• [Asterisk-Users] PRI E1 - HDLC Bad FCS / HDLC Abort errors   Steve Totaro
• [Asterisk-Users] New Product ID.   Steve Totaro
• [Asterisk-Users] Unable to prevent SIP to SIP calls from removing Asterisk from Media path   Johann
• [Asterisk-Users] Synthesized Voice for Asterisk   Steve Totaro
• [Asterisk-Users] PRI E1 - HDLC Bad FCS / HDLC Abort errors   Zoa
• [Asterisk-Users] mpg123 on x86_64 (Opteron MP)   Carlos Chavez
• [Asterisk-Users] ASTCC/ASTCC anything wrong with that?   bbench at mail.bg
• [Asterisk-Users] Re: Problem with Speex   Bharath
• [Asterisk-Users] Digium PCI-X timeline   Kevin P. Fleming
• [Asterisk-Users] Mechanisms for Implementing a Common ContactDatabase   Douglas Garstang
• [Asterisk-Users] Production Upgrades   Don Pobanz
• [Asterisk-Users] Outgoing data call   CSD
• [Asterisk-Users] PRI E1 - HDLC Bad FCS / HDLC Abort errors   Dushyanth Harinath
• [Asterisk-Users] Problems with current chan-capi-cm   Faris Raouf
• [Asterisk-Users] New Product ID.   C F
• [Asterisk-Users] uniqueid with multiple asterisk hosts   a0305292 at unet.univie.ac.at
• [Asterisk-Users] [helpp] Problem in astersik   Jaime Lopez
• [Asterisk-Users] Cisco 7940 Reboot   Aaron Daniel
• [Asterisk-Users] Re: Digium PCI-X timeline   Steven
• [Asterisk-Users] executing a reload under stress in Asterisk   Franklin Webb
• [Asterisk-Users] Attack dialing   Matt
• [Asterisk-Users] executing a reload under stress in Asterisk   Matt
• [Asterisk-Users] Zultys ZIP2 + asterisk + DTMF on other end? (i.e. ivrs, autoattendants, etc)   Dan Elder
• [Asterisk-Users] Need advice on BRI   Pedro Nunes
• [Asterisk-Users] Need advice on BRI   Kristof Hardy
• [Asterisk-Users] capi incoming call timeout   Louis-David Mitterrand
• [Asterisk-Users] Make list of incoming and outgoing calls   Arik Funke
• [Asterisk-Users] Make list of incoming and outgoing calls   Colin Anderson
• [Asterisk-Users] Make list of incoming and outgoing calls   Kristof Hardy
• [Asterisk-Users] PRI E1 - HDLC Bad FCS / HDLC Abort errors   Dushyanth Harinath
• [Asterisk-Users] Dlink DI-102 QOS Thingy?   Mojo Jojo
• [Asterisk-Users] trying to get SIP to work remotly.   Jason Brashear
• [Asterisk-Users] Linux Partitions (before asterisk install)   Johnny Voice
• [Asterisk-Users] trying to get SIP to work remotly.   Kerry Garrison
• [Asterisk-Users] Attack dialing   Wayne
• [Asterisk-Users] Call Monitoring / Ext to Ext with Sipura-841   Mike McMullen
• [Asterisk-Users] Make list of incoming and outgoing calls   Mojo with Horan & Company, LLC
• [Asterisk-Users] Linux Partitions (before asterisk install)   Tzafrir Cohen
• [Asterisk-Users] Linux Partitions (before asterisk install)   Matt Florell
• [Asterisk-Users] uniqueid with multiple asterisk hosts   Freddi Hansen
• [Asterisk-Users] PRI E1 - HDLC Bad FCS / HDLC Abort errors [ Solved]   Dushyanth Harinath
• [Asterisk-Users] Via Epia   Andrew Nowrot
• [Asterisk-Users] trying to get SIP to work remotly.   Ross C
• [Asterisk-Users] Attack dialing   Matt
• [Asterisk-Users] Linux Partitions (before asterisk install)   Kristian Kielhofner
• [Asterisk-Users] trying to get SIP to work remotly.   Mojo with Horan & Company, LLC
• [Asterisk-Users] Attack dialing   Wayne
• [Asterisk-Users] Daily Reboot Script for Asterisk Question   Mojo with Horan & Company, LLC
• [Asterisk-Users] Cisco 7940 Reboot   Kristian Kielhofner
• [Asterisk-Users] Cisco 7940 Reboot   Joseph
• [Asterisk-Users] Turning off hardware echo can on TE411P   Eric Bishop
• [Asterisk-Users] CDR MySQL   Juanjo Portela
• [Asterisk-Users] Cisco 7941 difference   Phil Genera
• [Asterisk-Users] asterisk in real estate developments   Brad Pauly
• [Asterisk-Users] CDR MySQL   tracinet
• [Asterisk-Users] asterisk in real estate developments   Colin Anderson
• [Asterisk-Users] busypattern tones?   Steven
• [Asterisk-Users] Cisco 7941 difference   Patrick
• [Asterisk-Users] Cisco 7941 difference   Cory Andrews
• [Asterisk-Users] Turning off hardware echo can on TE411P   Kevin P. Fleming
• [Asterisk-Users] asterisk in real estate developments   Steve Totaro
• [Asterisk-Users] Cisco 7940 Reboot   Aaron Daniel
• [Asterisk-Users] IP Phone Recommendation   Duracom ISP Lists
• [Asterisk-Users] Cisco 7940 Reboot   Patrick
• [Asterisk-Users] IP Phone Recommendation   Kerry Garrison
• [Asterisk-Users] Is clearglobalvars=no really working in asterisk 1.2.1?   Ulrich Holeschak
• [Asterisk-Users] IP Phone Recommendation   Anders Svensson
• [Asterisk-Users] Cisco 7940 Reboot   Kristian Kielhofner
• [Asterisk-Users] Linux Partitions (before asterisk install)   Pete Barnwell
• [Asterisk-Users] Cisco 7940 Reboot   Aaron Daniel
• [Asterisk-Users] IP Phone Recommendation   Mojo with Horan & Company, LLC
• [Asterisk-Users] ASTCC/ASTCC anything wrong with that?   Darren Wiebe
• [Asterisk-Users] Softphone with Hint support?   Avi Miller
• [Asterisk-Users] NOTIFY Messages   Douglas Garstang
• [Asterisk-Users] E1 PRI cause codes   Cameron Grant
• [Asterisk-Users] analog FXS card for dialup modem and fax   Isaac Xiao (KVB Kunlun Pty Limited)
• [Asterisk-Users] Interesting Article on Echo Cancellation   Cory Andrews
• [Asterisk-Users] IP Phone Recommendation   Ian Cowley
• [Asterisk-Users] new asterisk 1.2 setup doesn't react when I press any numbers   Brian
• [Asterisk-Users] Sip behind the NAT   janvb at caselaboratories.com
• [Asterisk-Users] asterisk in real estate developments   Brad Pauly
• [Asterisk-Users] Softphone with Hint support?   Kevin P. Fleming
• [Asterisk-Users] CDR MySQL   Juanjo Portela
• [Asterisk-Users] (Montreal Users) Call for technical presentations   Adrien Laurent
• [Asterisk-Users] CDR MySQL   Patrick
• [Asterisk-Users] CDR MySQL   Robert Webb
• [Asterisk-Users] Skips and Pops in Call Recordings   Matt Roth
• [Asterisk-Users] CDR MySQL   KRTorio
• [Asterisk-Users] Sipura + Asterisk 1.2 + dtmf   Asterisk User
• [Asterisk-Users] Invoking blind SIP REFER transfer asterisk   Karunakar Chemudugunta
• [Asterisk-Users] CDR MySQL   Patrick
• [Asterisk-Users] Sipura + Asterisk 1.2 + dtmf   C F
• [Asterisk-Users] No outgoing sound...sometimes   Jason Frisch
• [Asterisk-Users] Skips and Pops in Call Recordings   Matt Florell
• [Asterisk-Users] CDR MySQL   tracinet
• [Asterisk-Users] Sipura + Asterisk 1.2 + dtmf   tracinet
• [Asterisk-Users] Cisco 7940 Reboot   Rich Adamson
• [Asterisk-Users] Sipura + Asterisk 1.2 + dtmf   Asterisk User
• [Asterisk-Users] No outgoing sound...sometimes   Jason Frisch
• [Asterisk-Users] Sipura + Asterisk 1.2 + dtmf   Asterisk User
• [Asterisk-Users] Turning off hardware echo can on TE411P   William K. Volkman
• [Asterisk-Users] No outgoing sound...sometimes   Andrew Kohlsmith
• [Asterisk-Users] X100p echo guide   From PH
• [Asterisk-Users] Fax Service   list
• [Asterisk-Users] new asterisk 1.2 setup doesn't react when I press any numbers   KRTorio
• [Asterisk-Users] No outgoing sound...sometimes   Jason Frisch
• [Asterisk-Users] Spandsp   MBIT Technologies
• [Asterisk-Users] Small / embedded system recommendations   janvb at caselaboratories.com
• [Asterisk-Users] CDR MySQL   Innocent Evil
• [Asterisk-Users] Skips and Pops in Call Recordings   Colin Anderson
• [Asterisk-Users] No outgoing sound...sometimes   Rich Adamson
• [Asterisk-Users] Fax Service   C F
• [Asterisk-Users] X100p echo guide   Rich Adamson
• [Asterisk-Users] Small / embedded system recommendations   John Reynolds
• [Asterisk-Users] Patch to zaptel Makefile   Karl O. Pinc
• [Asterisk-Users] subscribing   asteriskmail
• [Asterisk-Users] NAT Issues?   scott
• [Asterisk-Users] Fax Service   George Pajari
• [Asterisk-Users] Realtime Subscribecontext   Kevin P. Fleming
• [Asterisk-Users] Patch zaptel.init to support debian   Karl O. Pinc
• [Asterisk-Users] No outgoing sound...sometimes   Jason Frisch
• [Asterisk-Users] Patch zaptel.init to support debian   Tzafrir Cohen
• [Asterisk-Users] ENUM For Presence   Douglas Garstang
• [Asterisk-Users] Small / embedded system recommendations   Chris Mason (Lists)
• [Asterisk-Users] Setting Language   René Enskat [Teamware GmbH]
• [Asterisk-Users] AGI GET Variable Problem   Kenige Ho
• [Asterisk-Users] IP Phone Recommendation   Kristof Hardy
• [Asterisk-Users] Cisco 7940 Reboot   Sergio Chersovani
• [Asterisk-Users] SPA-3000: Dual Registrations?   Brian Capouch
• [Asterisk-Users] Setting Language   Sergio Chersovani
• [Asterisk-Users] IP Phone Recommendation   Anders Svensson
• [Asterisk-Users] SPA-3000: Dual Registrations?   Kristian Kielhofner
• [Asterisk-Users] chan_capi AVM C2   stéphane plichon
• [Asterisk-Users] Re: CDR MySQL   Tomislav Parcina
• [Asterisk-Users] Re: AGI GET Variable Problem   Kenige Ho
• [Asterisk-Users] Info request from Sangoma users   Gavin Hamill
• [Asterisk-Users] chan_capi AVM C2   Philipp von Klitzing
• [Asterisk-Users] Softphone with Hint support?   Philipp von Klitzing
• [Asterisk-Users] Call Disconnecting   Code Lover
• [Asterisk-Users] queue_log Vs show queue abandon calls discrepancy   Dov Bigio
• [Asterisk-Users] OOH323 -> IAX2 : no sound   Eugene Prokopiev
• [Asterisk-Users] No outgoing sound...sometimes   Mario Evangelista-Silva
• [Asterisk-Users] Dlink DI-102 QOS Thingy?   Rob Hillis
• [Asterisk-Users] ericsson pabx and digium card TE110P   Henning Kilset Pedersen
• [Asterisk-Users] Info request from Sangoma users   Matt Florell
• [Asterisk-Users] g729 translation to zap (ISDN) doesn´t work   Klaus Peras
• [Asterisk-Users] IAX2 show channels show Channel (NONE)   Dmitry Zhukovski
• [Asterisk-Users] IAX2 show channels show Channel (NONE)   Zoa
• [Asterisk-Users] IAX2 show channels show Channel (NONE)   Dmitry Zhukovski
• [Asterisk-Users] NAT/Qualify/RTP bug   Arthur B Olsen
• [Asterisk-Users] IAX2 show channels show Channel (NONE)   Francesco Peeters (Asterisk)
• [Asterisk-Users] Info request from Sangoma users   Gavin Hamill
• [Asterisk-Users] Re: CDR MySQL   Tomislav Parcina
• [Asterisk-Users] SPA-3000: Dual Registrations?   Rich Adamson
• [Asterisk-Users] CDR MySQL   tracinet
• [Asterisk-Users] Sip behind the NAT   Michael George
• [Asterisk-Users] Patch zaptel.init to support debian   Kevin P. Fleming
• [Asterisk-Users] Info request from Sangoma users   Patrick
• [Asterisk-Users] ENUM For Presence   Kevin P. Fleming
• [Asterisk-Users] 1.2.1 has broken voicemail realtime switching   Joseph Rothstein
• [Asterisk-Users] No outgoing sound...sometimes   Andrew Kohlsmith
• [Asterisk-Users] No outgoing sound...sometimes   Andrew Kohlsmith
• [Asterisk-Users] Small / embedded system recommendations   Andrew Kohlsmith
• [Asterisk-Users] 1.2.1 has broken voicemail realtime switching   Kevin P. Fleming
• [Asterisk-Users] CDR MySQL   Juanjo Portela
• [Asterisk-Users] Turning off hardware echo can on TE411P   Jason Brashear
• [Asterisk-Users] Turning off hardware echo can on TE411P   Kevin P. Fleming
• [Asterisk-Users] Turning off hardware echo can on TE411P   Jason Brashear
• [Asterisk-Users] calls forwarded to busy agent   Patrick Fortin
• Re: [Asterisk-Users] g729 translation to zap (ISDN) doesn´t work   Klaus Peras
• [Asterisk-Users] SNOM 190 using 2 lines   Jason Brashear
• [Asterisk-Users] Polycom 501 remapping keys   Bill Gibbs
• [Asterisk-Users] Testing 10.0.0.203 with 10.0.0.0   Tomislav Parcina
• [Asterisk-Users] Info request from Sangoma users   Gavin Hamill
• [Asterisk-Users] extensions and regular expressions ( probablyan easy question )   Benjamin Lawetz
• [Asterisk-Users] RE: 1.2.1 has broken voicemail realtime   Joseph Rothstein
• [Asterisk-Users] Partial PRI pass thru?   Matt Burleigh
• [Asterisk-Users] Partial PRI pass thru?   Kevin P. Fleming
• [Asterisk-Users] RE: 1.2.1 has broken voicemail realtime   Aaron Daniel
• [Asterisk-Users] 408 Request Timeout vs. 403 Forbidden   Joseph Rothstein
• [Asterisk-Users] FXOTUNE Error on channel 2   Tom Vile
• [Asterisk-Users] SPA-3000: Dual Registrations?   Kerry Garrison
• [Asterisk-Users] Asterisk book feedback   Ross C
• [Asterisk-Users] calls forwarded to busy agent   Lenz
• [Asterisk-Users] Partial PRI pass thru?   Steve Totaro
• [Asterisk-Users] extension seen as busy when it is not   James Armstrong
• [Asterisk-Users] Asterisk book feedback   Austin Denyer
• [Asterisk-Users] Re: Odd DTMF issue over PRI   Steven
• [Asterisk-Users] Partial PRI pass thru?   Matt Burleigh
• [Asterisk-Users] Info request from Sangoma users   Matt Florell
• [Asterisk-Users] chan_capi AVM C2   stéphane plichon
• [Asterisk-Users] Re: Odd DTMF issue over PRI   Steven
• [Asterisk-Users] mISDN Caller ID problem   Pedro Nunes
• [Asterisk-Users] Skips and Pops in Call Recordings   Matt Roth
• [Asterisk-Users] SPA-3000: Dual Registrations?   Paul Hayes
• [Asterisk-Users] Asterisk book feedback   John Biundo
• [Asterisk-Users] Bonded ethernet ports and *   Rolf Brusletto
• [Asterisk-Users] talking about : mISDN Caller ID problem   Vladimir Montealegre
• [Asterisk-Users] Partial PRI pass thru?   Christian Victor
• [Asterisk-Users] Asterisk book feedback   Zoa
• [Asterisk-Users] chan_capi AVM C2   stéphane plichon
• [Asterisk-Users] Skips and Pops in Call Recordings   Matt Florell
• [Asterisk-Users] Patch zaptel.init to support debian   Karl O. Pinc
• [Asterisk-Users] Asterisk book feedback   Karl O. Pinc
• [Asterisk-Users] Asterisk Feature Request: app_bridgeme   Nir Simionovich - CTO
• [Asterisk-Users] Tellabs manuals   C F
• [Asterisk-Users] Asterisk book feedback   Jason Becker
• [Asterisk-Users] mISDN & chan_misdn on Fedora Core 4 - problems   Derek Conniffe
• [Asterisk-Users] Very high memory consumption when high number of calls are processed   Jon Bruel
• [Asterisk-Users] chan_capi AVM C2   Armin Schindler
• [Asterisk-Users] CID name & number contain unwanted quotes in CDR   Technical Support
• [Asterisk-Users] cdr_addon_mysql can't find libmysqlclient.so   Alejandro Mejia Evertsz
• [Asterisk-Users] WIFI Phones   rossi.tek
• [Asterisk-Users] Skips and Pops in Call Recordings   Matt Roth
• [Asterisk-Users] queues & music on hold   Dov Bigio
• [Asterisk-Users] Tellabs manuals   James Armstrong
• [Asterisk-Users] Tellabs manuals   C F
• [Asterisk-Users] Channel 0/1, span 1 got hangup request   Anton Bakulev
• [Asterisk-Users] format_mp3 & uninstalling mpg123   Dov Bigio
• [Asterisk-Users] Skips and Pops in Call Recordings   Matt Florell
• [Asterisk-Users] format_mp3 & uninstalling mpg123   Gil Kloepfer
• [Asterisk-Users] SPA-3000: Dual Registrations?   Brian Capouch
• [Asterisk-Users] Question on having asterisk put calls into a meetme.   Matt
• [Asterisk-Users] format_mp3 & uninstalling mpg123   Dov Bigio
• [Asterisk-Users] Partial PRI pass thru?   Rob Lith
• [Asterisk-Users] Meetme Conference sound problems   Alberto Risco
• [Asterisk-Users] IAX error message   Jonathan k. Creasy
• [Asterisk-Users] md 3200   Vladimir Montealegre
• [Asterisk-Users] format_mp3 & uninstalling mpg123   Gil Kloepfer
• [Asterisk-Users] pb !Astrisk 1.2 Card TE411P   michael delvoye
• [Asterisk-Users] pb !Astrisk 1.2 Card TE411P   Vladimir Montealegre
• [Asterisk-Users] pb !Astrisk 1.2 Card TE411P   michael delvoye
• [Asterisk-Users] mISDN & chan_misdn on Fedora Core 4 - problems   Avi Miller
• [Asterisk-Users] Re: Odd DTMF issue over PRI   Steven
• [Asterisk-Users] December VON Magazine   Brian Capouch
• [Asterisk-Users] empty line sip_notify.conf   Daren Pereira
• [Asterisk-Users] Asterisk 1.2.1   Leah Newmark
• [Asterisk-Users] Entering Digits   Leah Newmark
• [Asterisk-Users] chan_capi AVM C2   Michael J. Tubby G8TIC
• [Asterisk-Users] Asterisk 1.2.1   Justin Tunney
• [Asterisk-Users] IAXmodem, help   Miguel Soto
• [Asterisk-Users] IAXmodem, help   Lee Howard
• [Asterisk-Users] Bonded ethernet ports and *   Rich Adamson
• [Asterisk-Users] Small / embedded system recommendations   Chris Bagnall
• [Asterisk-Users] nice -n 19 called from shell script through Syst em() gives "Permission Denied"   Colin Anderson
• [Asterisk-Users] Skips and Pops in Call Recordings - channel.c Analysis   Matt Roth
• [Asterisk-Users] Skips and Pops in Call Recordings - channel.c Analysis   Adrian Carter
• [Asterisk-Users] Bonded ethernet ports and *   Watkins, Bradley
• [Asterisk-Users] X100p echo guide   Rich Adamson
• [Asterisk-Users] i found new website www.digiumchina.com in china   hoowa sun
• [Asterisk-Users] December VON Magazine   Rehan Ahmed
• [Asterisk-Users] Bonded ethernet ports and *   trixter aka Bret McDanel
• [Asterisk-Users] Asterisk 1.2 SIP register problems   Griffith, Michael
• [Asterisk-Users] asterisk in real estate developments   Chris Bagnall
• [Asterisk-Users] i found new website www.digiumchina.com in china   trixter aka Bret McDanel
• [Asterisk-Users] Bonded ethernet ports and *   Julio Arruda
• [Asterisk-Users] asterisk in real estate developments   David Phelan
• [Asterisk-Users] VizuFon CIP-4500 with Asterisk through SIP   isamar at isamarmaia.org
• [Asterisk-Users] outgoing calls that last an unreasonably long time   Adam Goryachev
• [Asterisk-Users] Dialing analog extensions from SIP?   Robert La Ferla
• [Asterisk-Users] Pattern Matching, speed and memory....   Rushowr
• [Asterisk-Users] December VON Magazine   Steve Totaro
• [Asterisk-Users] asterisk in real estate developments   Steve Totaro
• [Asterisk-Users] asterisk in real estate developments   Carlos
• [Asterisk-Users] New GSM 1-8 ports Gateway / Terminal forsale(with SMS Feature and Many more)   Sam Tam
• [Asterisk-Users] New GSM 1-8 ports Gateway / Terminal for sale(with SMS Feature and Many more)   Sam Tam
• [Asterisk-Users] asterisk in real estate developments   Colin Anderson
• [Asterisk-Users] Sip behind the NAT   Tom Rymes
• [Asterisk-Users] asterisk in real estate developments   Robert Augustyn
• [Asterisk-Users] Cisco 7960/ATA/MultiTech MVP200 FXS/FXO to H323 gateways on ebay   Tracy R Reed
• [Asterisk-Users] SIP Subscription Storage Location   Douglas Garstang
• [Asterisk-Users] fxs woes...   saad
• [Asterisk-Users] asterisk in real estate developments   Colin Anderson
• [Asterisk-Users] Dialing analog extensions from SIP?   Francesco Peeters (Asterisk)
• [Asterisk-Users] SIP Subscription Storage Location   Brian Capouch
• [Asterisk-Users] Long and variable echo   James Andrewartha
• [Asterisk-Users] RealTime and automatic extension registration.   Kristian Larsson
• [Asterisk-Users] chan_capi AVM C2   stéphane plichon
• [Asterisk-Users] send SMS via own SMS Service   Mrvka Andreas
• [Asterisk-Users] chan_capi AVM C2   Armin Schindler
• [Asterisk-Users] SIP peer vs. user-- how is the USER ever selected?   Steve Murphy
• [Asterisk-Users] send SMS via own SMS Service   Alejandro Alfonso
• [Asterisk-Users] chan_capi AVM C2   stéphane plichon
• [Asterisk-Users] chan_capi AVM C2   Armin Schindler
• [Asterisk-Users] PSTN gateway & Asterisk -> Virtual Switchboard???   Rafael Ledesma
• [Asterisk-Users] capi.conf - AVM C4 P2P or P2MP   Peer Oliver Schmidt
• [Asterisk-Users] chan_capi AVM C2   stéphane plichon
• [Asterisk-Users] TE410P and SPANDSP   Antonio Rabena
• [Asterisk-Users] Join when empty problem, in queue   Xavier Gil
• [Asterisk-Users] chan_capi AVM C2   Armin Schindler
• [Asterisk-Users] capi.conf - AVM C4 P2P or P2MP   Armin Schindler
• [Asterisk-Users] Re: extension seen as busy when it is not   Tomislav Parcina
• [Asterisk-Users] Wildcard TDM2400P: comments   yusuf
• [Asterisk-Users] TE410P and SPANDSP   Ma Zhiyong
• [Asterisk-Users] format_mp3 & uninstalling mpg123   Alessio Focardi
• [Asterisk-Users] Wildcard TDM2400P: comments   Jacques Leisy
• [Asterisk-Users] capi.conf - AVM C4 P2P or P2MP   Peer Oliver Schmidt
• [Asterisk-Users] voicemail boxes   Vinod
• [Asterisk-Users] Dial multiple destinations   Morten Tryfoss
• [Asterisk-Users] voicemail boxes   Dakota
• [Asterisk-Users] Join when empty problem, in queue   Morten Tryfoss
• [Asterisk-Users] Re: [helpp] Problem in astersik   Talat Ishtiaq
• [Fwd: Re: [Asterisk-Users] Re: [helpp] Problem in astersik]   Talat Ishtiaq
• [Asterisk-Users] voicemail boxes   Vinod
• [Asterisk-Users] capi.conf - AVM C4 P2P or P2MP   Armin Schindler
• [Asterisk-Users] Dial multiple destinations   a0305292 at unet.univie.ac.at
• [Asterisk-Users] [help] problem in astersik   Talat Ishtiaq
• [Asterisk-Users] Patch zaptel.init to support debian   Tzafrir Cohen
• [Asterisk-Users] Help:asterisk 1.2.1 release compile   julien bos
• [Asterisk-Users] Dial multiple destinations   Morten Tryfoss
• [Asterisk-Users] quadbri, isnd, netherlands: callerid not working   Mark Huizer
• [Asterisk-Users] Unable to find key   Alejandro Vargas
• [Asterisk-Users] extension seen as busy when it is not   Rich Adamson
• [Asterisk-Users] subscription   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Exceptionally long queue in SIP Channel   Aaron Clauson
• [Asterisk-Users] SIP Subscription Storage Location   BJ Weschke
• [Asterisk-Users] Asterisk 1.2.1   Tzafrir Cohen
• [Asterisk-Users] Bonded ethernet ports and *   Rolf Brusletto
• [Asterisk-Users] voicemail boxes   Don Pobanz
• [Asterisk-Users] IP Phone Recommendation   Sean Kennedy
• [Asterisk-Users] '#' (fast foward) and '*' (Rewind) not working in VoicemailMain   Chuck Bunn
• [Asterisk-Users] Unable to find key   James Armstrong
• [Asterisk-Users] fax and voice   hgaillac-sip at yahoo.fr
• [Asterisk-Users] MGCP Unable to find key   Steve Totaro
• [Asterisk-Users] voicemail boxes   Steve Totaro
• [Asterisk-Users] Need help with sipura   Talkvoip Telecom Canada
• [Asterisk-Users] Video calls (MS Messenger, Tandberg)   Jens.Kammann at dlr.de
• [Asterisk-Users] Need help with Sipura 3000   Talkvoip Telecom Canada
• [Asterisk-Users] Hint Priority for Polycom Phones   gw at adcomcorp.com
• [Asterisk-Users] Bonded ethernet ports and *   Rich Adamson
• [Asterisk-Users] Re: Testing 10.0.0.203 with 10.0.0.0   Tomislav Parcina
• [Asterisk-Users] Asterisk as client to PortaBilling   Andreas Mavrides
• [Asterisk-Users] Partial PRI pass thru?   Christian Victor
• [Asterisk-Users] md 3200   Tzafrir Cohen
• [Asterisk-Users] Need help with sipura   Rich Adamson
• [Asterisk-Users] Blind transferred user does not hear phone ring while waiting for phone to be picked up.   Chuck Bunn
• [Asterisk-Users] Partial PRI pass thru?   Steve Totaro
• [Asterisk-Users] Unable to find key   Philipp von Klitzing
• [Asterisk-Users] cdr_addon_mysql can't find libmysqlclient.so   Philipp von Klitzing
• [Asterisk-Users] Best way to automatically stop and start Asterisk nightly   Chuck Bunn
• [Asterisk-Users] Gateway crashes when transferring to external lines   Dustin Wenz
• [Asterisk-Users] Blind transferred user does not hear phone ring while waiting for phone to be picked up.   Chuck Bunn
• [Asterisk-Users] hardware echo cancellation for TDM card   Patrick Fortin
• [Asterisk-Users] RE: Asterisk to Avaya IP Office   David Rahn
• [Asterisk-Users] appradius   Juan Salas
• [Asterisk-Users] Join when empty problem, in queue   Diyanat Ali
• [Asterisk-Users] T1 to T1 dialout problem   Bart Fisher
• [Asterisk-Users] WIFI Phones   Matt Riddell
• [Asterisk-Users] OT: Linux on treo 650   C F
• [Asterisk-Users] Need help with sipura   C F
• [Asterisk-Users] hardware echo cancellation for TDM card   BJ Weschke
• [Asterisk-Users] hardware echo cancellation for TDM card   Kevin P. Fleming
• [Asterisk-Users] Polycom 501 remapping keys   Matthew T. O'Connor
• [Asterisk-Users] Cisco 7940 Time Source   Aaron Daniel
• [Asterisk-Users] WIFI Phones   Mojo with Horan & Company, LLC
• [Asterisk-Users] Headset Phones?   Kurth Bemis
• [Asterisk-Users] WIFI Phones   trixter aka Bret McDanel
• [Asterisk-Users] Background() followed by Read - something wrong?   Michaël Gaudette
• [Asterisk-Users] traffic shaping   Jose Limeres
• [Asterisk-Users] voicemail boxes   Kristian Kielhofner
• [Asterisk-Users] sharing a line w/multiple extensions   Robert La Ferla
• [Asterisk-Users] Cisco 7940 Time Source   Tad Heckaman
• [Asterisk-Users] OT: Linux on treo 650   Kerry Garrison
• Re: [Asterisk-Users] g729 translation to zap (ISDN) doesn´t work   Klaus Peras
• [Asterisk-Users] Headset Phones?   Garrett Smith
• [Asterisk-Users] WIFI Phones   Klaus Peras
• [Asterisk-Users] asterisk + H323 + 723   Kanishka Somaratne
• [Asterisk-Users] traffic shaping   Colin Anderson
• [Asterisk-Users] sharing a line w/multiple extensions   C F
• {Scanned} Re[2]: [Asterisk-Users] format_mp3 & uninstalling mpg123   hamshack.info
• [Asterisk-Users] sharing a line w/multiple extensions   Robert La Ferla
• [Asterisk-Users] traffic shaping   Casey Boone
• [Asterisk-Users] traffic shaping   Sean Kennedy
• [Asterisk-Users] Headset Phones?   Sean Cook
• [Asterisk-Users] HOWOT transfer call from mobile back to extension?   support at sjobeck.com
• [Asterisk-Users] Best way to automatically stop and start Asterisk nightly   Tzafrir Cohen
• [Asterisk-Users] sharing a line w/multiple extensions   Sean Cook
• [Asterisk-Users] sharing a line w/multiple extensions   Robert La Ferla
• [Asterisk-Users] Background() followed by Read - something wrong?   Luki
• [Asterisk-Users] traffic shaping   Rushowr
• [Asterisk-Users] ChanIsAvail() and SIP   Scott Maier
• [Asterisk-Users] 1.2.1 Compile Error   Peder at NetworkOblivion
• [Asterisk-Users] Re: 1.2.1 Compile Error   Peder at NetworkOblivion
• [Asterisk-Users] HOWOT transfer call from mobile back to extension?   James Armstrong
• [Asterisk-Users] OT: Linux on treo 650   Rusty Dekema
• [Asterisk-Users] OT: Linux on treo 650   C F
• [Asterisk-Users] Prevent Logging "reload" verbose   Samy Antoun
• [Asterisk-Users] OT: Linux on treo 650   C F
• [Asterisk-Users] OT: Linux on treo 650   C F
• [Asterisk-Users] Bonded ethernet ports and *   Julio Arruda
• [Asterisk-Users] Astcc help   Insider KT
• [Asterisk-Users] ChanIsAvail() and SIP   Jose Solares
• [Asterisk-Users] Bonded ethernet ports and *   Rich Adamson
• [Asterisk-Users] Fax detected, but no fax extension   hgaillac-sip at yahoo.fr
• [Asterisk-Users] OT: Linux on treo 650   Terry H. Gilsenan
• [Asterisk-Users] Fax detected, but no fax extension   Colin Anderson
• [Asterisk-Users] OT: Linux on treo 650   Freddi Hansen
• [Asterisk-Users] "Context Picker" for interception and redirection   Jason Lixfeld
• [Asterisk-Users] How to disable sip Native bridge   Jean-François Rousseau
• [Asterisk-Users] sharing a line w/multiple extensions   kevin ling
• [Asterisk-Users] OT: Linux on treo 650   Rusty Dekema
• [Asterisk-Users] traffic shaping   Andrew Kohlsmith
• [Asterisk-Users] Problem with bridging SIP to OH323 and SIP to SIP: Bridge stops bridging   Lukas Macura
• [Asterisk-Users] RE: Wildcard TDM2400P: comments   Isaac Xiao (KVB Kunlun Pty Limited)
• [Asterisk-Users] Astcc help   Darren Wiebe
• [Asterisk-Users] How to disable sip Native bridge   Steve Totaro
• [Asterisk-Users] Executing dailplan cmd from manager API   Elwin Andriol
• [Asterisk-Users] "Context Picker" for interception and redirection   Jason Lixfeld
• [Asterisk-Users] about sip softphone on windows smartphone   hoowa sun
• [Asterisk-Users] ChanIsAvail() and SIP   Scott Maier
• [Asterisk-Users] ChanIsAvail() and SIP   Jose Solares
• [Asterisk-Users] "Context Picker" for interception and redirection   Jose Solares
• [Asterisk-Users] Pops in Call Recordings Tied to Dropped Audio in Calls   Matt Roth
• [Asterisk-Users] Executing dailplan cmd from manager API   Elwin Andriol
• [Asterisk-Users] sharing a line w/multiple extensions   Kevin P. Fleming
• [Asterisk-Users] How to disable sip Native bridge   Kevin P. Fleming
• [Asterisk-Users] Pops in Call Recordings Tied to Dropped Audio in Calls   Kevin P. Fleming
• [Asterisk-Users] sharing a line w/multiple extensions   Robert La Ferla
• [Asterisk-Users] sharing a line w/multiple extensions   Kevin P. Fleming
• [Asterisk-Users] sharing a line w/multiple extensions   Robert La Ferla
• [Asterisk-Users] How to disable sip Native bridge   Aaron Daniel
• [Asterisk-Users] sharing a line w/multiple extensions   Adrian Carter
• [Asterisk-Users] Polycom Software   Darrick Hartman
• [Asterisk-Users] OT: Linux on treo 650   Rusty Dekema
• [Asterisk-Users] How to disable sip Native bridge   Kevin P. Fleming
• [Asterisk-Users] Polycom Software   Kevin P. Fleming
• [Asterisk-Users] sharing a line w/multiple extensions   Kevin P. Fleming
• [Asterisk-Users] SIP Subscription Storage Location   Douglas Garstang
• [Asterisk-Users] WishList - Devices that are (probably) not available yet [OFF-TOPIC]   Dhennys Pestana
• [Asterisk-Users] why sql error in asterisk 1.2.1 with realtime with mysql 4.1.x   hoowa sun
• [Asterisk-Users] Asterisk & STUN   Code Lover
• [Asterisk-Users] Starting RTP with Dial and MusicOnHold   Aaron Clauson
• [Asterisk-Users] I don't want ilbc, i just want G.711   Jason Chan (jasonOfficial)
• [Asterisk-Users] 2 PBX linked via internet   Ugo Bellavance
• [Asterisk-Users] SPA-3000: Dual Registrations?   Brian Capouch
• [Asterisk-Users] busypattern tones?   steve at daviesfam.org
• [Asterisk-Users] fxs woes...   saad
• [Asterisk-Users] RE: Wildcard TDM2400P: comments   yusuf
• [Asterisk-Users] fxs woes... solved. but why!   saad
• [Asterisk-Users] Anyone with VIP-450   Jason Chan (jasonOfficial)
• [Asterisk-Users] Starting RTP with Dial and MusicOnHold   Pedro Nunes
• [Asterisk-Users] AoC (Advice of Charge)   Tomislav Parcina
• [Asterisk-Users] Fax detected, but no fax extension   hgaillac-sip at yahoo.fr
• [Asterisk-Users] QueueMetrics 1.0 rc 1 out today   Lenz
• [Asterisk-Users] Starting RTP with Dial and MusicOnHold   Aaron Clauson
• [Asterisk-Users] background music...   Esteban Maestre
• [Asterisk-Users] How to tell if Authenticate failed without using j in 1.2   George Pajari
• [Asterisk-Users] Help with mgcp   Alejandro Vargas
• [Asterisk-Users] again - show queue info   Dov Bigio
• [Asterisk-Users] 2 PBX linked via internet   Umair Bari
• [Asterisk-Users] I don't want ilbc, i just want G.711   Umair Bari
• [Fwd: Re: [Asterisk-Users] Re: [helpp] Problem in astersik]   Umair Bari
• [Asterisk-Users] How to change the Dial command H option to ## ?   Obelix
• [Asterisk-Users] Firewall Ports forward   Umair Bari
• [Asterisk-Users] SPA-3000: Dual Registrations?   Rich Adamson
• [Asterisk-Users] RE: Asterisk-Users Digest, Vol 17, Issue 89   Michaël Gaudette
• [Asterisk-Users] How to change the Dial command H option to ## ?   Umair Bari
• [Asterisk-Users] sharing a line w/multiple extensions   Rich Adamson
• [Asterisk-Users] RE: how to forward call within office   Tejas Shah
• [Asterisk-Users] oh323 : which versions recommended for asterisk 1.2?   Eugene Prokopiev
• [Asterisk-Users] Small explanation of txgain rxgain statement please   Steve Davies
• [Asterisk-Users] EXITWITHQUEUE on queue_log   Dov Bigio
• [Asterisk-Users] Firewall Ports forward   Pablo Allietti
• [Asterisk-Users] screen safe_asterisk does'nt spawn asterisk   Simone Cittadini
• [Asterisk-Users] sharing a line w/multiple extensions   Rich Adamson
• [Asterisk-Users] How to change the Dial command H option to ## ?   Matt Riddell
• [Asterisk-Users] RE: how to forward call within office   Steve Totaro
• [Asterisk-Users] chan-capi avm b1 and capi.conf problems   Ricardo
• [Asterisk-Users] hint on Zap channels   DRi at b-w-computer.de
• [Asterisk-Users] hint on Zap channels   Sergio Chersovani
• [Asterisk-Users] voicemail cutting out   simprix
• [Asterisk-Users] sharing a line w/multiple extensions   Walt Reed
• [Asterisk-Users] RE: how to forward call within office   Tomislav Parcina
• [Asterisk-Users] I don't want ilbc, i just want G.711   Elton Machado
• [Asterisk-Users] Starting RTP with Dial and MusicOnHold   Elton Machado
• [Asterisk-Users] Small explanation of txgain rxgain statement please   Rich Adamson
• [Asterisk-Users] Re: 2 PBX linked via internet   Ugo Bellavance
• [Asterisk-Users] Sip configuration for make and receive calls   asterisk at barnatech.com
• [Asterisk-Users] ChanIsAvail()   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Re: 2 PBX linked via internet   Ugo Bellavance
• [Asterisk-Users] Re: voicemail cutting out   Tomislav Parcina
• [Asterisk-Users] RE: Wildcard TDM2400P: comments   Tom Vile
• [Asterisk-Users] ChanIsAvail()   Jose Solares
• [Asterisk-Users] RE: Wildcard TDM2400P: comments   Kevin P. Fleming
• [Asterisk-Users] hint on Zap channels   DRi at b-w-computer.de
• [Asterisk-Users] ChanIsAvail()   Alexander Lopez
• [Asterisk-Users] Starting RTP with Dial and MusicOnHold   Aaron Clauson
• [Asterisk-Users] function cut()   Steve Hanselman
• [Asterisk-Users] background music...   Matt
• [Asterisk-Users] astcc issue   jonny hashem
• [Asterisk-Users] How to disable sip Native bridge   Jean-François Rousseau
• [Asterisk-Users] Fax detected, but no fax extension   Colin Anderson
• [Asterisk-Users] sharing a line w/multiple extensions   Robert La Ferla
• [Asterisk-Users] E1 Echo (was: Small explanation of txgain rxgain statement please)   Steve Davies
• [Asterisk-Users] sharing a line w/multiple extensions   Rich Adamson
• [Asterisk-Users] Pops in Call Recordings Tied to Dropped Audio in Calls   Matt Roth
• [Asterisk-Users] Pops in Call Recordings Tied to Dropped Audio in Calls   Kevin P. Fleming
• [Asterisk-Users] function cut()   Steve Hanselman
• [Asterisk-Users] E1 Echo (was: Small explanation of txgain rxgain statement please)   Rich Adamson
• [Asterisk-Users] Outbound Routing   Brendan Mannella
• [Asterisk-Users] Can you time limit access to a trunk?   David Tillman
• [Asterisk-Users] function cut()   Steve Hanselman
• [Asterisk-Users] traffic shaping   Andrew Kohlsmith
• [Asterisk-Users] sharing a line w/multiple extensions   John Biundo
• [Asterisk-Users] Shutting down Asterisk when not in RTP Stream   Douglas Garstang
• [Asterisk-Users] Pops in Call Recordings Tied to Dropped Audio in Calls   Matt Roth
• [Asterisk-Users] Handyton 486 Outbound problem   Craig Bruenderman
• [Asterisk-Users] Shutting down Asterisk when not in RTP Stream   Douglas Garstang
• [Asterisk-Users] Re: chan-capi avm b1 and capi.conf problems   Stefan Tichy
• [Asterisk-Users] Voipsupply - my experience   Terry H. Gilsenan
• [Asterisk-Users] E1 Echo (was: Small explanation of txgain rx gain statement please)   Colin Anderson
• [Asterisk-Users] Shutting down Asterisk when not in RTP Stream   Diyanat Ali
• [Asterisk-Users] Google Analytics and voip-info.org   Denis Galvão - iSolve
• [Asterisk-Users] How to change the Dial command H option to ## ?   Obelix
• [Asterisk-Users] Originating calls to a channel groups   asterisk at barnatech.com
• [Asterisk-Users] RE: HELP!   Matt Schulte
• [Asterisk-Users] Asterisk Realtime connection failed   Insider KT
• [Asterisk-Users] Sip behind the NAT   Michael George
• [Asterisk-Users] traffic shaping   Matt Riddell
• [Asterisk-Users] Voipsupply - my experience   Dennis Gilmore
• [Asterisk-Users] Shutting down Asterisk when not in RTP Stream   Douglas Garstang
• [Asterisk-Users] Google Analytics and voip-info.org   David K Parker
• [Asterisk-Users] Disposition Failed still happening   Aaron Daniel
• [Asterisk-Users] Google Analytics and voip-info.org   Colin Anderson
• [Asterisk-Users] traffic shaping   Matt Riddell
• [Asterisk-Users] Voipsupply - my experience   Omar A. Sabek
• [Asterisk-Users] Echo Canceller usage   Kevin P. Fleming
• [Asterisk-Users] Echo Canceller usage   Richard Scobie
• [Asterisk-Users] Disposition Failed still happening   tracinet
• [Asterisk-Users] Echo Canceller usage   Richard Scobie
• [Asterisk-Users] Disposition Failed still happening   tracinet
• [Asterisk-Users] sharing a line w/multiple extensions   Andrew Latham
• [Asterisk-Users] Voipsupply - my experience   Shaw Terwilliger
• [Asterisk-Users] Echo Canceller usage   Kevin P. Fleming
• [Asterisk-Users] Disposition Failed still happening   Aaron Daniel
• [Asterisk-Users] Re: chan-capi avm b1 and capi.conf problems   Avi Miller
• [Asterisk-Users] about g729   Martin Joseph
• [Asterisk-Users] Script to detect corrupted faxes from SpanDSP   Colin Anderson
• [Asterisk-Users] sharing a line w/multiple extensions   Rich Adamson
• [Asterisk-Users] Disposition Failed still happening   tracinet
• [Asterisk-Users] looking for hardphone configuration info   Bruce Ferrell
• [Asterisk-Users] Will ooh323 ever move from addons?   Rich Adamson
• [Asterisk-Users] looking for hardphone configuration info   Philip Edelbrock
• [Asterisk-Users] Re: looking for hardphone configuration info   Steven
• [Asterisk-Users] sharing a line w/multiple extensions   Robert La Ferla
• [Asterisk-Users] sharing a line w/multiple extensions   snacktime
• [Asterisk-Users] Connecting Meridian M8x24-DS to Asterisk - No DTMF tones   Philip Edelbrock
• [Asterisk-Users] Will ooh323 ever move from addons?   Kevin P. Fleming
• [Asterisk-Users] looking for hardphone configuration info   Cory Andrews
• [Asterisk-Users] E1 Echo   Florian Overkamp
• [Asterisk-Users] Weird IAX trunking/7960/ILBC quality issue   Chris Bagnall
• [Asterisk-Users] Weird IAX trunking/7960/ILBC quality issue   Steve Totaro
• [Asterisk-Users] PRI_SWITCH_EUROISDN_T1   janvb at caselaboratories.com
• [Asterisk-Users] Asterisk & STUN   Steve Totaro
• [Asterisk-Users] Connecting Meridian M8x24-DS to Asterisk - No DTMFtones   Steve Totaro
• [Asterisk-Users] SIP Trunk please help   Ryan Pagquil
• [Asterisk-Users] Re: PROGRESS with cause code 31 received   Dana Olson
• [Asterisk-Users] Re: PROGRESS with cause code 31 received   Jonathan Augenstine
• [Asterisk-Users] Echo & TDM11B   David K Parker
• [Asterisk-Users] Connecting Meridian M8x24-DS to Asterisk - No DTMFtones   Philip Edelbrock
• [Asterisk-Users] getting started   sukrit
• [Asterisk-Users] getting started   Time Bandit
• [Asterisk-Users] Alternative source for Asterisk-IM   Takayuki Uehara
• [Asterisk-Users] Astcc help   JP Carballo
• [Asterisk-Users] Fax detected, but no fax extension   JP Carballo
• [Asterisk-Users] getting started   Diyanat Ali
• [Asterisk-Users] getting started   Chandan Mishra
• [Asterisk-Users] SIP Trunk please help   Diyanat Ali
• [Asterisk-Users] CallerID/Extension Matching with Realtime Extensions   Douglas Garstang
• [Asterisk-Users] SIP Trunk please help   Ryan Pagquil
• [Asterisk-Users] CallerID/Extension Matching with RealtimeExtensions   Douglas Garstang
• [Asterisk-Users] sharing a line w/multiple extensions   Jean-Michel Hiver
• [Asterisk-Users] Raltime database schemas   Douglas Garstang
• [Asterisk-Users] SIP Trunk please help   Diyanat Ali
• [Asterisk-Users] Calls not incoming to any extension fromremoteproxy server   saad
• [Asterisk-Users] Alternative source for Asterisk-IM   kevin ling
• [Asterisk-Users] CallerID/Extension Matching with RealtimeExtensions   Brian Capouch
• [Asterisk-Users] Weird IAX trunking/7960/ILBC quality issue   Chris Bagnall
• [Asterisk-Users] hint on Zap channels   Sergio Chersovani
• [Asterisk-Users] Weird IAX trunking/7960/ILBC quality issue   Chris Bagnall
• [Asterisk-Users] asterisk 1.2 mysql cdr garbage   Kristof Hardy
• [Asterisk-Users] Fax detected, but no fax extension   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Meetme option Ax   tcchan
• [Asterisk-Users] hint on Zap channels   DRi at b-w-computer.de
• [Asterisk-Users] Configuration of two Asterisk server   Mantu Jha
• [Asterisk-Users] Alternative source for Asterisk-IM   Mario Evangelista-Silva
• [Asterisk-Users] E1 Echo (was: Small explanation of txgain rx gain statement please)   Steve Davies
• [Asterisk-Users] Raltime database schemas   BJ Weschke
• [Asterisk-Users] Does hardware like this exist...?   Evert Meulie
• [Asterisk-Users] HW Echo Cancellers   Jason Kim
• [Asterisk-Users] Romania/Rumania setup   FaberK
• [Asterisk-Users] How to change the Dial command H option to ## ?   Obelix
• [Asterisk-Users] Will ooh323 ever move from addons?   Rich Adamson
• [Asterisk-Users] E1 Echo   Rich Adamson
• [Asterisk-Users] ztdummy / timer problem with kernel 2.6.14   Fredrik Emil Jensen
• [Asterisk-Users] cdr mysql problem   Mohammad Shokuie
• [Asterisk-Users] Does hardware like this exist...?   BJ Weschke
• [Asterisk-Users] Connecting Meridian M8x24-DS to Asterisk - No DTMFtones   Rich Adamson
• [Asterisk-Users] cdr mysql problem   Doug Lytle
• [Asterisk-Users] cdr mysql problem   Christian Victor
• [Asterisk-Users] Re: AoC (Advice of Charge)   Tomislav Parcina
• [Asterisk-Users] getting started   Rich Adamson
• [Asterisk-Users] Central Registration mechanism   Noel Athaide
• [Asterisk-Users] E1 Echo   Florian Overkamp
• [Asterisk-Users] cdr mysql problem   Diyanat Ali
• [Asterisk-Users] cdr mysql problem   Rich Adamson
• [Asterisk-Users] Re: Meetme option Ax   Tony Mountifield
• [Asterisk-Users] incoming dtmf handling by ATA devices ?   Luigi Rizzo
• [Asterisk-Users] E1 Echo   Rich Adamson
• [Asterisk-Users] Experience sharing on Planet VIP-450 + Asterisk   Jason Chan (jasonOfficial)
• [Asterisk-Users] E1 Echo   Andrew Kohlsmith
• [Asterisk-Users] incoming dtmf handling by ATA devices ?   Rich Adamson
• [Asterisk-Users] Re: Does hardware like this exist...?   Steven
• [Asterisk-Users] E1 Echo   Florian Overkamp
• [Asterisk-Users] E1 Echo   Florian Overkamp
• [Asterisk-Users] Re: Connecting Meridian M8x24-DS to Asterisk - NoDTMFtones   Steven
• [Asterisk-Users] Experience sharing on Planet VIP-450 + Asterisk   Steve Totaro
• [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with ringing   Michael J. Tubby B.Sc (Hons) G8TIC
• [Asterisk-Users] Connecting Meridian M8x24-DS to Asterisk - NoDTMFtones   Steve Totaro
• [Asterisk-Users] E1 Echo   Andrew Kohlsmith
• [Asterisk-Users] Re: ztdummy / timer problem with kernel 2.6.14   Tomislav Parcina
• [Asterisk-Users] A rather big setup.   Mike Fedyk
• [Asterisk-Users] Re: AoC (Advice of Charge)   Frank Sautter
• [Asterisk-Users] HW Echo Cancellers   Darren Wright
• [Asterisk-Users] E1 Echo   Rich Adamson
• [Asterisk-Users] Mediatrix 1204 help please.   Ariel Batista
• [Asterisk-Users] Mediatrix 1204 help please.   Nathan C. Smith
• [Asterisk-Users] CallerID/Extension Matching with Realtime Extensions   Kevin P. Fleming
• [Asterisk-Users] hint on Zap channels   Kevin P. Fleming
• [Asterisk-Users] Re: Does hardware like this exist...?   Evert Meulie
• [Asterisk-Users] E1 Echo   Florian Overkamp
• [Asterisk-Users] Mediatrix 1204 help please.   Rich Adamson
• [Asterisk-Users] hint on Zap channels   C F
• [Asterisk-Users] hint on Zap channels   DRi at b-w-computer.de
• [Asterisk-Users] hint on Zap channels   Kevin P. Fleming
• [Asterisk-Users] Re: Does hardware like this exist...?   Kevin P. Fleming
• [Asterisk-Users] Shutting down Asterisk when not in RTP Stream   dashy dude
• [Asterisk-Users] HW Echo Cancellers   Steve Davies
• [Asterisk-Users] about g729   ram
• [Asterisk-Users] 1.2.0 queue.conf exit context   Jason Lixfeld
• [Asterisk-Users] HW Echo Cancellers   Darren Wright
• [Asterisk-Users] Having trouble calling out from Zap channel   Michael Sampson
• [Asterisk-Users] 1.2.0 queue.conf exit context   Kevin P. Fleming
• [Asterisk-Users] 1.2.0 queue.conf exit context   Jason Lixfeld
• [Asterisk-Users] HW Echo Cancellers   Steve Davies
• [Asterisk-Users] Music On Hold   Bud Bach
• [Asterisk-Users] Asterisk won´t load module codec_g729a.so   Klaus Peras
• [Asterisk-Users] Music On Hold   Klaus Peras
• [Asterisk-Users] Music On Hold   Bud Bach
• [Asterisk-Users] Music On Hold   Bud Bach
• [Asterisk-Users] CID lookup from an Exchange Public folder   Steve Hanselman
• [Asterisk-Users] Amtelco Infinity   Michael Sampson
• [Asterisk-Users] Digium TE205 Card   San Singhania
• [Asterisk-Users] CID lookup from an Exchange Public folder   Colin Anderson
• [Asterisk-Users] Aastra 480i   Michael George
• [Asterisk-Users] hint on Zap channels   Francesco Peeters (Asterisk)
• [Asterisk-Users] Experience sharing on Planet VIP-450 + Asterisk   JP Carballo
• [Asterisk-Users] .call files on PRI not waiting for answer in de sired context   Colin Anderson
• [Asterisk-Users] Merlin Legend mode codes   Sean Cook
• [Asterisk-Users] hint on Zap channels   Kevin P. Fleming
• [Asterisk-Users] .call files on PRI not waiting for answer in de sired context   Kevin P. Fleming
• [Asterisk-Users] hint on Zap channels   Francesco Peeters (Asterisk)
• [Asterisk-Users] .call files on PRI not waiting for answer in desired context   Colin Anderson
• [Asterisk-Users] Aastra 480i   Carlos Chavez
• [Asterisk-Users] .call files on PRI not waiting for answer in de sired context <--ResponseTimeout the best answer?   Colin Anderson
• [Asterisk-Users] FOP button limit?   Kerry Garrison
• [Asterisk-Users] FOP button limit?   Terry H. Gilsenan
• [Asterisk-Users] asterisk + H323 + 723   Kanishka Somaratne
• [Asterisk-Users] Asterisk-1.2.1 incomplete DID number on incoming T1 line   Bob Lichtenberg
• [Asterisk-Users] asterisk + H323 + 723   Alberto Sagredo
• [Asterisk-Users] Central Registration mechanism   Douglas Garstang
• [Asterisk-Users] IAX Jitterbuffer and trunking   Steve Kann
• [Asterisk-Users] Aastra 480i   Dave
• [Asterisk-Users] Central Registration mechanism   Aaron Daniel
• [Asterisk-Users] Central Registration mechanism   Douglas Garstang
• [Asterisk-Users] FOP button limit?   Kerry Garrison
• [Asterisk-Users] Aastra 480i   Carlos Chavez
• [Asterisk-Users] FOP button limit?   Terry H. Gilsenan
• [Asterisk-Users] DIGIUM   VOICEIN at aol.com
• [Asterisk-Users] DIGIUM   Kevin P. Fleming
• [Asterisk-Users] A2billing Trunk   MapsAir
• [Asterisk-Users] Aastra 480i   Robert La Ferla
• [Asterisk-Users] DIGIUM   Ross C
• [Asterisk-Users] A2billing Trunk   snacktime
• [Asterisk-Users] Codecs.   Rich Adamson
• [Asterisk-Users] Re: DIGIUM   Tony Mountifield
• [Asterisk-Users] DIGIUM   gsmith at voipsupply.com
• [Asterisk-Users] DIGIUM   trixter aka Bret McDanel
• [Asterisk-Users] DIGIUM   Ross C
• [Asterisk-Users] TDM01B answering issue   chawki hammoud
• [Asterisk-Users] Asterisk Redundancy   Malik Muhammad
• [Asterisk-Users] Codecs.   Pablo Allietti
• [Asterisk-Users] Re: Codecs.   Pablo Allietti
• [Asterisk-Users] Redency of Asterisk   Steve
• [Asterisk-Users] Redency of Asterisk   Steve Totaro
• [Asterisk-Users] Redency of Asterisk   kevin ling
• [Asterisk-Users] A2billing Trunk   ram
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   Hiu Yen Onn
• [Asterisk-Users] IAX Jitterbuffer and trunking   Richard Scobie
• [Asterisk-Users] FOP button limit?   Jerry Jones
• [Asterisk-Users] FOP button limit?   Kerry Garrison
• [Asterisk-Users] Asterisk & STUN   Code Lover
• [Asterisk-Users] A2billing Trunk   Lan
• [Asterisk-Users] A2billing Trunk   Rob Hillis
• [Asterisk-Users] asterisk and h323 problems   Kanishka Somaratne
• [Asterisk-Users] Asterisk & STUN   Niklas Larsson
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   Matt Florell
• [Asterisk-Users] Best way to automatically stop and start Asterisk nightly   Jean-Michel Hiver
• [Asterisk-Users] Can Asterisk replace Cisco 5350?   Linuxnizer The Mesmorizer
• [Asterisk-Users] Can Asterisk replace Cisco 5350?   Jean-Michel Hiver
• [Asterisk-Users] Re: Does hardware like this exist...?   Erwin de Raad
• [Asterisk-Users] CID lookup from an Exchange Public folder   Steve Hanselman
• [Asterisk-Users] Can Asterisk replace Cisco 5350?   AR Tarzi
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   Simone Cittadini
• [Asterisk-Users] Can Asterisk replace Cisco 5350?   Jean-Michel Hiver
• [Asterisk-Users] I need syntax on applicationmap in features.conf   Obelix
• [Asterisk-Users] screen safe_asterisk does'nt spawn asterisk   Tzafrir Cohen
• [Asterisk-Users] Can Asterisk replace Cisco 5350?   Linuxnizer The Mesmorizer
• [Asterisk-Users] Key R (Flash) and Asterisk   Linux Administator
• [Asterisk-Users] Can Asterisk replace Cisco 5350?   Jean-Michel Hiver
• [Asterisk-Users] .call files on PRI not waiting for answer in de sired context <--ResponseTimeout the best answer?   Philipp von Klitzing
• [Asterisk-Users] asterisk 1.2.1 realtime mysql.4.1.xx report errors   hoowa sun
• [Asterisk-Users] Re: Codecs.   Rich Adamson
• [Asterisk-Users] A2billing Trunk   Steve Totaro
• [Asterisk-Users] TDM01B answering issue   Rich Adamson
• [Asterisk-Users] Cid_rewrite update   Technical Support
• [Asterisk-Users] FOP button limit?   Nicolás Gudiño
• [Asterisk-Users] multiple ALSA devices and Asterisk   Dan
• [Asterisk-Users] Re: [Astguiclient-users] [PATCH][RFC] Quiet debugging messages in Net::MySQL Perl module   Mike Fedyk
• [Asterisk-Users] Cisco 79xx display as busy-lamp field   Bruce Komito
• [Asterisk-Users] Strange problem with sjphone and 1.2.1   Evil Skymarshal
• [Asterisk-Users] Alarm panel through ATA   Chris Mason (Lists)
• [Asterisk-Users] i can't register to my sip service(but x-lite can)   hoowa sun
• [Asterisk-Users] Strange problem with sjphone and 1.2.1   Chuck Bunn
• [Asterisk-Users] Alarm panel through ATA   Rich Adamson
• [Asterisk-Users] Alarm panel through ATA   Gonzalo Gonzalez
• [Asterisk-Users] Strange problem with sjphone and 1.2.1   Evil Skymarshal
• [Asterisk-Users] Strange problem with sjphone and 1.2.1   Chuck Bunn
• [Asterisk-Users] Strange problem with sjphone and 1.2.1   Chuck Bunn
• [Asterisk-Users] Strange problem with sjphone and 1.2.1   Rich Adamson
• [Asterisk-Users] Strange problem with sjphone and 1.2.1   Chuck Bunn
• [Asterisk-Users] Terminating calls externally via SER   Douglas Garstang
• [Asterisk-Users] Alarm panel through ATA   Jerry Jones
• [Asterisk-Users] CID lookup from an Exchange Public folder   Colin Anderson
• [Asterisk-Users] Linksys PAP2 and Asterisk   Jason (WeatherServer)
• [Asterisk-Users] Patch zaptel.init to support debian   Kevin P. Fleming
• [Asterisk-Users] Grandstream GXP-2000 Auto Answer   William M. Sandiford
• [Asterisk-Users] MusicOnHold not working   Jason Lixfeld
• [Asterisk-Users] Linux Partitions (before asterisk install)   Michiel van Baak
• [Asterisk-Users] TDM01B answering issue   chawki hammoud
• [Asterisk-Users] Best way to automatically stop and start Asterisk nightly   Chuck Bunn
• [Asterisk-Users] Linux Partitions (before asterisk install)   Andrew Kohlsmith
• [Asterisk-Users] placing a call in one or several call groups   hgaillac-sip at yahoo.fr
• [Asterisk-Users] TDM01B answering issue   Andrew Kohlsmith
• [Asterisk-Users] Sangoma E1 board Experience   janvb at caselaboratories.com
• [Asterisk-Users] Sangoma E1 board Experience   Andrew Kohlsmith
• [Asterisk-Users] Sangoma E1 board Experience   janvb at caselaboratories.com
• [Asterisk-Users] Linux Partitions (before asterisk install)   Tzafrir Cohen
• [Asterisk-Users] TDM01B answering issue   chawki hammoud
• [Asterisk-Users] Linksys PAP2 and Asterisk   John Biundo
• [Asterisk-Users] Can't pickup call when dialing *8 extension   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Linux Partitions (before asterisk install)   Michiel van Baak
• [Asterisk-Users] Linux Partitions (before asterisk install)   Michiel van Baak
• [Asterisk-Users] Teliax billing question   Ryan Burke
• [Asterisk-Users] Grandstream GXP-2000 Auto Answer   Julian J. M.
• [Asterisk-Users] Re: Teliax billing question   Wolfgang S. Rupprecht
• [Asterisk-Users] Toll Free Providers   Tom Vile
• [Asterisk-Users] Can't pickup call when dialing *8 extension   Rich Adamson
• [Asterisk-Users] Teliax billing question   Rich Adamson
• [Asterisk-Users] Re: Teliax billing question   Ryan Burke
• [Asterisk-Users] SIP and echo cancel   Mike Bernson
• [Asterisk-Users] Re: Teliax billing question   Rich Adamson
• [Asterisk-Users] Sangoma E1 board Experience   Rich Adamson
• [Asterisk-Users] aastra.cfg & mac.cfg examples Firmware version 1.3   Lists
• [Asterisk-Users] Grandstream GXP-2000 Auto Answer   Rob Thomas
• [Asterisk-Users] TDM01B answering issue   Eric "ManxPower" Wieling
• [Asterisk-Users] TDM01B answering issue   Andrew Kohlsmith
• [Asterisk-Users] Re: Teliax billing question   Ryan Burke
• [Asterisk-Users] TDM01B answering issue   Steve Underwood
• [Asterisk-Users] hint on Zap channels   C F
• [Asterisk-Users] HW Echo Cancellers   C F
• [Asterisk-Users] HW Echo Cancellers   C F
• [Asterisk-Users] indications.conf for Japan?   Robert La Ferla
• [Asterisk-Users] Mediatrix 1204 help please.   C F
• [Asterisk-Users] ztdummy problem !!!   Gabriel Sartor
• [Asterisk-Users] SIP and echo cancel   Luki
• [Asterisk-Users] ztdummy problem !!!   Luki
• [Asterisk-Users] screen safe_asterisk does'nt spawn asterisk   Simone Cittadini
• [Asterisk-Users] TDM01B answering issue   chawki hammoud
• [Asterisk-Users] PERL AGI DIALSTATUS   Code Lover
• [Asterisk-Users] Toll Free Providers   Rehan Ahmed
• [Asterisk-Users] Can't pickup call when dialing *8 extension   hgaillac-sip at yahoo.fr
• [Asterisk-Users] cdr mysql problem   Mohammad Shokuie
• [Asterisk-Users] Can't pickup call when dialing *8 extension (resent)   hgaillac-sip at yahoo.fr
• [Asterisk-Users] asterisk 1.2.1 and mixmonitor problem   Mohammad Shokuie
• [Asterisk-Users] Too high volume on Music on Hold   Alberto Sagredo
• [Asterisk-Users] ztdummy problem !!!   Insider KT
• [Asterisk-Users] ztdummy problem !!!   Tzafrir Cohen
• [Asterisk-Users] Is this possible in Asterisk?   Christian
• [Asterisk-Users] ztdummy problem !!!   Doug Lytle
• [Asterisk-Users] Is this possible in Asterisk?   Doug Lytle
• [Asterisk-Users] Is this possible in Asterisk?   Christian
• [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with ringing   Armin Schindler
• [Asterisk-Users] Asterisk <-> Avaya system   Kristian Larsson
• [Asterisk-Users] astcc issue   Darren Wiebe
• [Asterisk-Users] Is this possible in Asterisk?   Elmar Haneke
• [Asterisk-Users] Is this possible in Asterisk?   Andrew Kohlsmith
• [Asterisk-Users] Shutting down Asterisk when not in RTP Stream   Douglas Garstang
• [Asterisk-Users] SIP Watchdog   Mike Hammett
• [Asterisk-Users] New voicemail alert options for Cisco 7960 SIP phones   Tim Connolly
• [Asterisk-Users] Is this possible in Asterisk?   Christian
• [Asterisk-Users] ACD with polycom ip phones   hgaillac-sip at yahoo.fr
• [Asterisk-Users] SIP Watchdog   turby at seznam.cz
• [Asterisk-Users] Is it me, or is 1.2.1 slower than 1.0.9?   Francesco Peeters (Asterisk)
• [Asterisk-Users] New voicemail alert options for Cisco 7960 SIPphones   Kerry Garrison
• [Asterisk-Users] ztdummy problem !!!   C F
• [Asterisk-Users] Is this possible in Asterisk?   Mohammad Shokuie
• [Asterisk-Users] SIP and echo cancel   Mohammad Shokuie
• [Asterisk-Users] Is this possible in Asterisk?   Andrew Kohlsmith
• [Asterisk-Users] SIP and echo cancel   Mohammad Shokuie
• [Asterisk-Users] TDM01B answering issue   Mohammad Shokuie
• [Asterisk-Users] Too high volume on Music on Hold   Jason Lixfeld
• [Asterisk-Users] SIP and echo cancel   Andrew Kohlsmith
• [Asterisk-Users] TDM01B answering issue   chawki hammoud
• [Asterisk-Users] Sangoma E1 board Experience   janvb at caselaboratories.com
• [Asterisk-Users] FOP led Colors   Alex Montoanelli
• [Asterisk-Users] Is it me, or is 1.2.1 slower than 1.0.9?   Francesco Peeters (Asterisk)
• [Asterisk-Users] iaxmodem through zaphfc   Massimo De Nadal
• [Asterisk-Users] Sangoma E1 board Experience   Rich Adamson
• [Asterisk-Users] Sangoma E1 board Experience   Andrew Kohlsmith
• [Asterisk-Users] Sangoma E1 board Experience   Steve Underwood
• [Asterisk-Users] Extension processing misunderstanding   Bradley Schatz
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   Hiu Yen Onn
• [Asterisk-Users] Sangoma E1 board Experience   Rich Adamson
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   Hiu Yen Onn
• [Asterisk-Users] Configuration of two Asterisk server   JP Carballo
• [Asterisk-Users] Anybody having trouble terminating calls at Voxee? <eom>   Tom Lynn
• [Asterisk-Users] Odd problem with Encore 201-SA (r2 converter) with asterisk   Fernando Romo
• [Asterisk-Users] Odd problem with Encore 201-SA (r2 converter) with asterisk   Fernando Romo
• [Asterisk-Users] [Fwd: Odd problem with Encore 201-SA (r2 converter) with asterisk]   Fernando Romo
• [Asterisk-Users] SIP Remote Call Control   Jason Kim
• [Asterisk-Users] Asterisk Voice mail-reg   nr k
• [Asterisk-Users] Recording Volume on Zap Channel   Gulzar Hussain
• [Asterisk-Users] SIP and echo cancel   Philip Edelbrock
• [Asterisk-Users] sharing a line w/multiple extensions   Adam Goryachev
• [Asterisk-Users] SIP Trunk please help   Ryan Pagquil
• [Asterisk-Users] Configuration of two Asterisk server   JP Carballo
• [Asterisk-Users] Anybody having trouble terminating calls at Voxee? <eom>   JP Carballo
• [Asterisk-Users] Re: Does hardware like this exist...?   Evert Meulie
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Evert Meulie
• [Asterisk-Users] Re: Best way to automatically stop and start Asterisk nightly   Tomislav Parcina
• [Asterisk-Users] Junk at the beginning of frame   Dov Bigio
• [Asterisk-Users] Looking for a spare LCD display SNOM 220   Loek Gijben
• [Asterisk-Users] SIP and echo cancel   Cirelle Enterprises
• [Asterisk-Users] DTMFMODE with grandstream   giti at dataproducts.ae
• [Asterisk-Users] SIP and echo cancel   Andrew Kohlsmith
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Francesco Peeters (Asterisk)
• [Asterisk-Users] Re: Codecs.   Pablo Allietti
• [Asterisk-Users] ACD with polycom ip phones (resent)   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Re: Codecs.   Rich Adamson
• [Asterisk-Users] [Fwd: Odd problem with Encore 201-SA (r2 converter) with asterisk]   Steve Underwood
• [Asterisk-Users] CID lookup from an Exchange Public folder   Steve Hanselman
• [Asterisk-Users] Shutting down Asterisk when not in RTP Stream   BJ Weschke
• [Asterisk-Users] Asterisk Limitations   James Sturges
• [Asterisk-Users] ACD with polycom ip phones   Matthew
• [Asterisk-Users] DTMFMODE with grandstream   Rich Adamson
• [Asterisk-Users] ACD with polycom ip phones   Kevin P. Fleming
• [Asterisk-Users] ACD with polycom ip phones   hgaillac-sip at yahoo.fr
• [Asterisk-Users] ACD with polycom ip phones   Adam Goryachev
• [Asterisk-Users] ACD with polycom ip phones   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Callware VoiceOne released: a new, easy web GUI   kleis-asterisk-dev at tiscali.it
• [Asterisk-Users] ACD with polycom ip phones   Kevin P. Fleming
• [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with ringing   Jason Williams
• [Asterisk-Users] NVFaxDetect   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Can't call out on ZAP channel - need help   Michael Sampson
• [Asterisk-Users] ACD with polycom ip phones   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Paul Hewlett
• [Asterisk-Users] DTMFMODE with grandstream   giti at dataproducts.ae
• [Asterisk-Users] Problem using Queue and Sip Soft   Julien SIRBU
• [Asterisk-Users] Can't call out on ZAP channel - need help   O'Connor, Jonathan
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   Walt Reed
• [Asterisk-Users] Can't call out on ZAP channel - need help   Michael Sampson
• [Asterisk-Users] Re: ztdummy / timer problem with kernel 2.6.14   Fredrik Emil Jensen
• [Asterisk-Users] unsubscribe please   Jason Brashear
• [Asterisk-Users] Can't call out on ZAP channel - need help   O'Connor, Jonathan
• [Asterisk-Users] Can't call out on ZAP channel - need help   Michael Sampson
• [Asterisk-Users] Can't pass variables using Originate in PHPAGI 2.14   Anish Basu
• [Asterisk-Users] Can't call out on ZAP channel - need help   O'Connor, Jonathan
• [Asterisk-Users] MixMonitor error exit   asterisk183
• [Asterisk-Users] iax2 on a server behind a linux based stateful firewall   Sean Kennedy
• {Scanned} [Asterisk-Users] Asterisk Voice mail-reg   hamshack.info
• {Scanned} [Asterisk-Users] aastra.cfg & mac.cfg examples Firmware version 1.3   hamshack.info
• [Asterisk-Users] PERL AGI DIALSTATUS   Apu Islam
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 111   Lawrence B Thaler
• [Asterisk-Users] Can't call out on ZAP channel - need help   Michael Sampson
• [Asterisk-Users] Re: Teliax billing question   Wolfgang S. Rupprecht
• [Asterisk-Users] problem with automatic attender calls   Xavier Gil
• [Asterisk-Users] Variable Help   Johnathan Falk
• [Asterisk-Users] Re: Teliax billing question   trixter aka Bret McDanel
• [Asterisk-Users] DTMFMODE with grandstream   Rich Adamson
• [Asterisk-Users] Re: Does hardware like this exist...?   Sam Tam
• [Asterisk-Users] iax2 on a server behind a linux based stateful firewall   Rich Adamson
• [Asterisk-Users] asterisk 1.2.1 and mixmonitor problem   Maximiliano J. Goldsmid
• [Asterisk-Users] Distinctive Ring and zapata.conf   Robert La Ferla
• [Asterisk-Users] asterisk 1.2.1 and mixmonitor problem   imran ahmed
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Mark Hulber
• [Asterisk-Users] Originate a call to a Queue?   Jim Miller
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 111   AR Tarzi
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Brian Capouch
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Kerry Garrison
• [Asterisk-Users] VoIP/VPN providers in Switzerland   Juan Jose Comellas
• [Asterisk-Users] Simulate incoming line   Chris Mason (Lists)
• [Asterisk-Users] Re: Codecs.   Pablo Allietti
• [Asterisk-Users] Asterisk Limitations   pdhales at optusnet.com.au
• [Asterisk-Users] Mulitple voicemail on mulitple phones   kurt x
• [Asterisk-Users] IBM eServers?   Harry McGregor
• [Asterisk-Users] Mulitple voicemail on mulitple phones   Kerry Garrison
• [Asterisk-Users] Simulate incoming line   C F
• [Asterisk-Users] Sangoma E1 board Experience   David Yat Sin
• [Asterisk-Users] IBM eServers?   Chad Osmond
• [Asterisk-Users] queues and redirection.   Peter Ankerstål
• [Asterisk-Users] hangup detection   Jonathan
• [Asterisk-Users] Simulate incoming line   Chris Mason (Lists)
• [Asterisk-Users] VoIP/VPN providers in Switzerland   Asterisk
• [Asterisk-Users] Asterisk & NAT behaviour   Guenther Starnberger
• [Asterisk-Users] Strange problem with sjphone and 1.2.1   Evil Skymarshal
• [Asterisk-Users] ser --> sip.conf --->extensions.conf, variable context   Scott
• [Asterisk-Users] A2billing Trunk   Maps
• [Asterisk-Users] hangup detection   Diego Andrés Asenjo González
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   AR Tarzi
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Kerry Garrison
• [Asterisk-Users] ALERT_INFO Not Working Upon Upgrade to 1.2.1   Nikhil Yogesh Jogia
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Luigi Rizzo
• [Asterisk-Users] Originate a call to a Queue?   lenz
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Kerry Garrison
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Michiel van Baak
• [Asterisk-Users] Handytone 486 Outbound problem   Craig Bruenderman
• [Asterisk-Users] ALERT_INFO Not Working Upon Upgrade to 1.2.1   Mojo with Horan & Company, LLC
• [Asterisk-Users] ALERT_INFO Not Working Upon Upgrade to 1.2.1   BJ Weschke
• [Asterisk-Users] Handytone 486 Outbound problem   Chris Bagnall
• [Asterisk-Users] Handytone 486 Outbound problem   Jerry Jones
• [Asterisk-Users] ALERT_INFO Not Working Upon Upgrade to 1.2.1   BJ Weschke
• [Asterisk-Users] Handling SIP clients behind NAT on a semi-dynamic IP   Chris Bagnall
• [Asterisk-Users] IBM eServers?   Jolly M. Recto
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Luigi Rizzo
• [Asterisk-Users] IBM eServers?   BJ Weschke
• [Asterisk-Users] Polycom retry interval and DNS SRV failover   Anthony Rodgers
• [Asterisk-Users] IBM eServers?   Ilan Rabinovitch
• [Asterisk-Users] memory not being released   Bruno de Assumpção Loureiro
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   isamar at isamarmaia.org
• [Asterisk-Users] Affordable IP Phones for Asterisk   Dakota
• [Asterisk-Users] Affordable IP Phones for Asterisk   Julio Tejera
• [Asterisk-Users] IBM eServers?   Kevin P. Fleming
• [Asterisk-Users] ALERT_INFO Not Working Upon Upgrade to 1.2.1   Nikhil Yogesh Jogia
• [Asterisk-Users] Toll Free Providers   Michael Graves
• [Asterisk-Users] ALERT_INFO Not Working Upon Upgrade to 1.2.1   BJ Weschke
• [Asterisk-Users] IBM eServers?   BJ Weschke
• [Asterisk-Users] ACD with polycom ip phones   BJ Weschke
• [Asterisk-Users] Toll Free Providers   John Reynolds
• [Asterisk-Users] New voicemail alert options for Cisco 7960 SIP phones   C F
• [Asterisk-Users] Asterisk with Uniden uip200   Steven Job
• [Asterisk-Users] Asterisk with Uniden uip200   Waldo Rubinstein
• [Asterisk-Users] Asterisk with Uniden uip200   Waldo Rubinstein
• [Asterisk-Users] SIP - SIP bridge dropping calls?   Adrian A
• [Asterisk-Users] Asterisk with Uniden uip200   Steven Job
• [Asterisk-Users] Asterisk with Uniden uip200   Waldo Rubinstein
• [Asterisk-Users] SIP and echo cancel   Mike Bernson
• [Asterisk-Users] ACD with polycom ip phones   Douglas Garstang
• [Asterisk-Users] Asterisk with Uniden uip200   Steven Job
• [Asterisk-Users] To write Sphinx Interface in EAGI or app_xxx.c?   L
• [Asterisk-Users] UNREACHABLE PEER   scott
• [Asterisk-Users] PERL AGI DIALSTATUS   Paul Tinsley
• [Asterisk-Users] Digium TDM2400 Series Server Compatability   Cory Andrews
• [Asterisk-Users] Asterisk with Uniden uip200   Waldo Rubinstein
• [Asterisk-Users] Fast AGi Variables   Alberto Sagredo
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Kristian Larsson
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Jean-Michel Hiver
• [Asterisk-Users] music on hold problem   Matt
• [Asterisk-Users] music on hold problem   Fredrik Emil Jensen
• [Asterisk-Users] Re: Asterisk <-> Skype anywhere/anyhow?   Evert Meulie
• [Asterisk-Users] Re: Is it me, or is 1.2.1 slower than 1.0.9?   Tomislav Parcina
• [Asterisk-Users] Re: Problem using Queue and Sip Soft   Tomislav Parcina
• [Asterisk-Users] Re: Asterisk <-> Skype anywhere/anyhow?   Evert Meulie
• [Asterisk-Users] music on hold problem   Matt
• [Asterisk-Users] Re: Very high memory consumption when high number of calls are processed   Jon Brüel
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Tzafrir Cohen
• [Asterisk-Users] ACD with polycom ip phones   hgaillac-sip at yahoo.fr
• [Asterisk-Users] ACD with polycom ip phones   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Lenz
• [Asterisk-Users] messages of Mobile Operator   Matteo Piazza
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Tzafrir Cohen
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Luigi Rizzo
• [Asterisk-Users] HW Echo Cancellers   Steve Davies
• [Asterisk-Users] Re: New voicemail alert options for Cisco 7960 SIP phones   Ben Buxton
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Steve Kennedy
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Kristian Larsson
• [Asterisk-Users] inbound routing with amp and TDM400   asterisk at frameweb.it
• [Asterisk-Users] inbound routing with amp and TDM400   Rich Adamson
• [Asterisk-Users] ACD with polycom ip phones   BJ Weschke
• [Asterisk-Users] messages of Mobile Operator   Diego Ercolani
• [Asterisk-Users] UNREACHABLE PEER   Rich Adamson
• [Asterisk-Users] ACD with polycom ip phones   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Soporte   Will Velez
• [Asterisk-Users] How to get received digits from console channel   Phuong Nguyen
• [Asterisk-Users] Digium TDM2400 Series Server Compatability   Kevin P. Fleming
• [Asterisk-Users] Soporte   Erick Perez
• [Asterisk-Users] HW Echo Cancellers   Matt Florell
• [Asterisk-Users] music on hold problem   Dov Bigio
• [Asterisk-Users] 482 Loop Detected when transferring calls back to Asterisk   David Allen
• [Asterisk-Users] TE205P E1 PRI card and other problems   Antoine Megalla
• [Asterisk-Users] Rolling dialplan... best practice?   Ryan Booz
• [Asterisk-Users] Goto after Dial PRoblem   René Enskat [Teamware GmbH]
• [Asterisk-Users] Re: Is it me, or is 1.2.1 slower than 1.0.9?   Francesco Peeters (Asterisk)
• [Asterisk-Users] ACD with polycom ip phones   Douglas Garstang
• [Asterisk-Users] ACD with polycom ip phones   Douglas Garstang
• [Asterisk-Users] ACD with polycom ip phones   Douglas Garstang
• [Asterisk-Users] Fast AGi Variables   Moises Silva
• [Asterisk-Users] Variable Help   Moises Silva
• [Asterisk-Users] Handling SIP clients behind NAT on a semi-dynamic IP   Bharath
• [Asterisk-Users] Soporte   Guillermo Salas M
• [Asterisk-Users] Asterisk & Broadvoice help??   Shawn Porter
• [Asterisk-Users] Queues and Agents   Johann
• [Asterisk-Users] Re: Romania/Rumania setup   FaberK
• ot[Asterisk-Users] Soporte   Vladimir Montealegre
• [Asterisk-Users] Asterisk & Broadvoice help??   Bruce Komito
• [Asterisk-Users] Asterisk & Broadvoice help??   Steven Job
• [Asterisk-Users] iax2 on a server behind a linux based stateful firewall   Sean Kennedy
• [Asterisk-Users] SIP/IAX to PSTN   pat cito
• [Asterisk-Users] MOH engaged while holding for ANOTHER party (1.2.1)   Ronald Lewis
• [Asterisk-Users] Asterisk & Broadvoice help??   Shawn Porter
• [Asterisk-Users] 1.2.1 Queues   Douglas Garstang
• [Asterisk-Users] Re: Romania/Rumania setup   Chris HARIGA
• [Asterisk-Users] Asterisk & Broadvoice help??   Steven Job
• [Asterisk-Users] 1.2.1 Queues   Douglas Garstang
• [Asterisk-Users] How to get received digits from console channel   Jonathan k. Creasy
• [Asterisk-Users] Toll Free Providers   Erwin de Raad
• [Asterisk-Users] G729 and Cisco IOS 12.4   Todd Weiser
• [Asterisk-Users] SunFire X4100   Peder at NetworkOblivion
• [Asterisk-Users] meet me room status   Dov Bigio
• [Asterisk-Users] 1.2.1 Queues   Mojo with Horan & Company, LLC
• [Asterisk-Users] 1.2.1 Queues   Douglas Garstang
• [Asterisk-Users] IVR and db   Serge Schumacher
• [Asterisk-Users] 1.2.1 Queues   Mojo with Horan & Company, LLC
• [Asterisk-Users] Asterisk on Compact PCI platform   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Distinctive Ring and zapata.conf   Robert La Ferla
• [Asterisk-Users] PERL AGI DIALSTATUS   Furqan
• [Asterisk-Users] 1.2.1 Queues   Brian Capouch
• [Asterisk-Users] Asterisk & FXO & Panasonic PBX   Waldo Rubinstein
• [Asterisk-Users] Is this possible in Asterisk?   Furqan
• [Asterisk-Users] Toll Free Providers   Julio Tejera
• [Asterisk-Users] Is this possible in Asterisk?   Furqan
• [Asterisk-Users] Linking existing channels through Manager interface. Is it possible?   Senyo
• [Asterisk-Users] IVR and db   Jean-Michel Hiver
• [Asterisk-Users] Re: Mulitple voicemail on mulitple phones   Steven
• [Asterisk-Users] IVR and db   Serge Schumacher
• [Asterisk-Users] 1.2.1 Queues   Douglas Garstang
• [Asterisk-Users] Linking existing channels through Managerinterface. Is it possible?   Shawn Porter
• [Asterisk-Users] Analog terminals and modems? does it work   Chris Gamble
• [Asterisk-Users] Distinctive Ring and zapata.conf   Shawn Porter
• [Asterisk-Users] 1.2.1 Queues   Peter Bowyer
• [Asterisk-Users] 1.2.1 Queues   Aaron Daniel
• [Asterisk-Users] 3 Phone Call Qualtiy Issues   Rhonda Herron
• [Asterisk-Users] Analog terminals and modems? does it work   Bjorn Asmul
• [Asterisk-Users] Linking existing channels through Managerinterface. Is it possible?   Matt Florell
• [Asterisk-Users] Rolling dialplan... best practice?   pdhales at optusnet.com.au
• [Asterisk-Users] Asterisk & FXO & Panasonic PBX   Bjorn Asmul
• [Asterisk-Users] How do I remove the temp greeting?!?!   tracinet
• [Asterisk-Users] IVR Capacity   Serge Schumacher
• [Asterisk-Users] IVR Capacity   Chris Tooley
• [Asterisk-Users] RFC 3262 PRACK   Trond Andersen
• [Asterisk-Users] Asterisk & FXO & Panasonic PBX   Waldo Rubinstein
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Quark IT - Hilton Travis
• [Asterisk-Users] IVR Capacity   Shawn Porter
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] IVR Capacity   Shawn Porter
• [Asterisk-Users] IVR and db   Jean-Michel Hiver
• [Asterisk-Users] SIP Subscriptions   Peter Bowyer
• [Asterisk-Users] IVR Capacity   Jean-Michel Hiver
• [Asterisk-Users] IVR Capacity   Serge Schumacher
• [Asterisk-Users] IVR and db   Serge Schumacher
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] SIP Subscriptions   Peter Bowyer
• [Asterisk-Users] SIP Subscriptions   Darren Wiebe
• [Asterisk-Users] Help Debugging Dropped Call Audio   Matt Roth
• [Asterisk-Users] IVR and db   Jean-Michel Hiver
• [Asterisk-Users] IVR Capacity   snacktime
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] SIP Subscriptions   Gary Reuter
• [Asterisk-Users] SIP Subscriptions   Darren Wiebe
• [Asterisk-Users] SPANDSP & TX RX Fax paid support wanted.   Rehan Ahmed
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] IVR Capacity   Jean-Michel Hiver
• [Asterisk-Users] Re: Can you time limit access to a trunk?   Wolfgang S. Rupprecht
• [Asterisk-Users] RE: SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] SPANDSP & TX RX Fax paid support wanted.   Administrator TOOTAI
• [Asterisk-Users] RE: SIP Subscriptions   BJ Weschke
• [Asterisk-Users] Meetme and ztdummy   Douglas Garstang
• [Asterisk-Users] RE: SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] Meetme and ztdummy   BJ Weschke
• [Asterisk-Users] Got SUBSCRIBE for extensions without hint   Anthony Rodgers
• [Asterisk-Users] RE: SIP Subscriptions   BJ Weschke
• [Asterisk-Users] Got SUBSCRIBE for extensions without hint   Douglas Garstang
• [Asterisk-Users] RE: SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] Got SUBSCRIBE for extensions without hint   BJ Weschke
• [Asterisk-Users] Meetme and ztdummy   Douglas Garstang
• [Asterisk-Users] Affordable IP Phones for Asterisk   Rehan Ahmed
• [Asterisk-Users] Got SUBSCRIBE for extensions without hint   Anthony Rodgers
• [Asterisk-Users] RE: SIP Subscriptions   BJ Weschke
• [Asterisk-Users] Unicall E1 Error in Mexico   Jorge Cisneros
• [Asterisk-Users] IVR Capacity   Serge Schumacher
• [Asterisk-Users] Affordable IP Phones for Asterisk   Philip Edelbrock
• [Asterisk-Users] RE: SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] Latest Source   Douglas Garstang
• [Asterisk-Users] Latest Source   Tony Hoyle
• [Asterisk-Users] Latest Source   Douglas Garstang
• [Asterisk-Users] Asterisk Hardware recomendation   Hiu Yen Onn
• [Asterisk-Users] 1.2.1 Queues   Scott Plante
• [Asterisk-Users] Latest Source   Mojo with Horan & Company, LLC
• [Asterisk-Users] 1.2.1 Queues   C F
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   C F
• [Asterisk-Users] Latest Source   Steve Totaro
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   BJ Weschke
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   William Boehlke
• [Asterisk-Users] SIP Subscriptions   C F
• [Asterisk-Users] SIP Subscriptions   C F
• [Asterisk-Users] Asterisk based IVR/VoiceMail Server for a Unified Messaging suite   ammad jami
• [Asterisk-Users] SIP Subscriptions   C F
• [Asterisk-Users] Sangoma E1 board Experience   janvb at caselaboratories.com
• [Asterisk-Users] Asterisk based IVR/VoiceMail Server for a Unified Messaging suite   ammad jami
• [Asterisk-Users] Linking existing channels through Managerinterface. Is it possible?   C F
• [Asterisk-Users] Re: Mulitple voicemail on mulitple phones   C F
• [Asterisk-Users] Sangoma E1 board Experience   janvb at caselaboratories.com
• [Asterisk-Users] Linking existing channels through Managerinterface. Is it possible?   Matt Florell
• [Asterisk-Users] Sangoma E1 board Experience   janvb at caselaboratories.com
• [Asterisk-Users] SIP Subscriptions   pdhales at optusnet.com.au
• [Asterisk-Users] low audio volume on recorded .wav voicemail messages   Robert La Ferla
• [Asterisk-Users] SIP Subscriptions   Aaron Daniel
• [Asterisk-Users] Asterisk based IVR/VoiceMail Server for a Unified Messaging suite   pdhales at optusnet.com.au
• [Asterisk-Users] SIP Subscriptions   pdhales at optusnet.com.au
• [Asterisk-Users] IVR Capacity   janvb at caselaboratories.com
• [Asterisk-Users] Latest Source   Douglas Garstang
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] Latest Source   C F
• [Asterisk-Users] Latest Source   Douglas Garstang
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] Latest Source   C F
• [Asterisk-Users] SIP Subscriptions   Aaron Daniel
• [Asterisk-Users] SIP Subscriptions   Darren Wiebe
• [Asterisk-Users] Digium E1 Card Modprobe problems   Arinze Izukanne
• [Asterisk-Users] Latest Source   Douglas Garstang
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] Latest Source   Darren Wiebe
• [Asterisk-Users] dtmf problem   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] SIP Subscriptions   Aaron Daniel
• [Asterisk-Users] Latest Source   C F
• [Asterisk-Users] Latest Source   Aaron Daniel
• [Asterisk-Users] Latest Source   Douglas Garstang
• [Asterisk-Users] Latest Source   Aaron Daniel
• [Asterisk-Users] Latest Source   Douglas Garstang
• [Asterisk-Users] Asterisk and Adtran TA 750 Channel Bank -- odd behavior (help!)   Gaurav Naik
• [Asterisk-Users] SIP Subscriptions   Steve Totaro
• [Asterisk-Users] Latest Source   Steve Totaro
• [Asterisk-Users] SIP Subscriptions   C F
• [Asterisk-Users] Latest Source   Kevin P. Fleming
• [Asterisk-Users] RFC 3262 PRACK   Kevin P. Fleming
• [Asterisk-Users] How do I remove the temp greeting?!?!   Kevin P. Fleming
• [Asterisk-Users] Re: Mulitple voicemail on mulitple phones   Kevin P. Fleming
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] SIP Subscriptions   Jon Radon
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] Unicall Problem with fax   Jorge Cisneros
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   Rob Lith
• [Asterisk-Users] Busy recognition   Douglas Garstang
• [Asterisk-Users] Latest Source   Peter Bowyer
• [Asterisk-Users] Latest Source   Peter Bowyer
• [Asterisk-Users] Latest Source   Brian Capouch
• [Asterisk-Users] Latest Source   Douglas Garstang
• [Asterisk-Users] Latest Source   Douglas Garstang
• [Asterisk-Users] Latest Source   Douglas Garstang
• [Asterisk-Users] MFC/R2   Jorge Cisneros
• [Asterisk-Users] 3 Phone Call Qualtiy Issues   Martin Joseph
• [Asterisk-Users] dtmf problem   asterisk-users at ecessor.com
• [Asterisk-Users] Rolling dialplan... best practice?   Erik
• [Asterisk-Users] WG: Goto after Dial PRoblem   René Enskat [Teamware GmbH]
• [Asterisk-Users] hangup detection   steve at daviesfam.org
• [Asterisk-Users] Asterisk <-> Skype anywhere/anyhow?   steve at daviesfam.org
• [Asterisk-Users] 482 Loop Detected when transferring calls back to Asterisk   Olle E. Johansson
• [Asterisk-Users] 1.2.1 Queues   Olle E. Johansson
• [Asterisk-Users] RFC 3262 PRACK   Olle E. Johansson
• [Asterisk-Users] SIP Subscriptions   Olle E. Johansson
• [Asterisk-Users] RE: SIP Subscriptions   Olle E. Johansson
• [Asterisk-Users] RE: SIP Subscriptions   Olle E. Johansson
• [Asterisk-Users] Got SUBSCRIBE for extensions without hint   Olle E. Johansson
• [Asterisk-Users] SIP Subscriptions   Olle E. Johansson
• [Asterisk-Users] Re: SIP Subscriptions   Tony Mountifield
• [Asterisk-Users] Rolling dialplan... best practice?   Peer Oliver Schmidt
• [Asterisk-Users] Re: SIP Subscriptions   Tony Mountifield
• [Asterisk-Users] Re: SIP Subscriptions   Peter Bowyer
• [Asterisk-Users] ISDN   francesco giuliani
• [Asterisk-Users] Re: SIP Subscriptions   steve at daviesfam.org
• [Asterisk-Users] ISDN   Giovanni Miano
• [Asterisk-Users] Re: Asterisk & Broadvoice help??   Tomislav Parcina
• [Asterisk-Users] Re: music on hold problem   Tomislav Parcina
• [Asterisk-Users] Re: Rolling dialplan... best practice?   Tomislav Parcina
• [Asterisk-Users] Problem with CDR   Dan Journo
• [Asterisk-Users] MP3 problems: MP3Player and Musiconhold   Wolfgang Borgon
• [Asterisk-Users] Latest Source   Walt Reed
• [Asterisk-Users] RE: Asterisk & FXO & Panasonic PBX   Tomislav Parcina
• [Asterisk-Users] turn off message "Silence suppression ..." on Asterisk console   Eugene Prokopiev
• [Asterisk-Users] Latest Source   Rich Adamson
• [Asterisk-Users] Tracing a crash with CAPI calls   Andrew Gough
• [Asterisk-Users] Re: music on hold problem   Doug Lytle
• [Asterisk-Users] Sangoma E1 board Experience   Steve Underwood
• [Asterisk-Users] Re: music on hold problem   Doug Lytle
• [Asterisk-Users] Instalar Ubuntu   Will Velez
• [Asterisk-Users] Analog terminals and modems? does it work   Rich Adamson
• [Asterisk-Users] [offtopic] Asterisk <-IP-> Siemens HiPath 4000   Dmitry Ivanov
• [Asterisk-Users] ASTERISK CALL ROUTER   Dumpolid Exeplish
• [Asterisk-Users] Re: Re: AoC (Advice of Charge)   Tomislav Parcina
• [Asterisk-Users] Re: Re: music on hold problem   Tomislav Parcina
• [Asterisk-Users] Postgres   andrutto
• [Asterisk-Users] ISDN   Armin Schindler
• [Asterisk-Users] Tracing a crash with CAPI calls   Armin Schindler
• [Asterisk-Users] Asterisk server to provide virtuals IPBX   hgaillac-sip at yahoo.fr
• [Asterisk-Users] php agi problem (perhaps problem..)   asterisk at frameweb.it
• [Asterisk-Users] [offtopic] Asterisk <-IP-> Siemens HiPath 4000   bbench at mail.bg
• [Asterisk-Users] How do I remove the temp greeting?!?!   tracinet
• [Asterisk-Users] Asterisk Call Forwarding   Androtech
• [Asterisk-Users] Asterisk server to provide virtuals IPBX   Dmitry Ivanov
• [Asterisk-Users] Sangoma E1 board Experience   janvb at caselaboratories.com
• [Asterisk-Users] php agi problem (perhaps problem..)   trixter aka Bret McDanel
• [Asterisk-Users] Re: Re: RFC 3262 PRACK (Olle E. Johansson)   Trond Andersen
• [Asterisk-Users] Using mgcp get/generate "message waiting indication"   in out
• [Asterisk-Users] show queue   Dov Bigio
• [Asterisk-Users] 3 Phone Call Qualtiy Issues   Rhonda Herron
• [Asterisk-Users] Asterisk Call Forwarding   Alberto Sagredo
• [Asterisk-Users] Calls not incoming to any extension from remote proxy server   bbench at mail.bg
• [Asterisk-Users] [offtopic] Asterisk <-IP-> Siemens HiPath 4000   richard Coco
• [Asterisk-Users] port vs bindport   Tomislav Parcina
• [Asterisk-Users] dtmf problem   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Asterisk Call Forwarding   Time Bandit
• [Asterisk-Users] Polycom 500 IP and problems with show hints   Kib Eki
• [Asterisk-Users] Unicall E1 Error in Mexico   Martinez Felix
• [Asterisk-Users] Asterisk Call Forwarding   Androtech
• [Asterisk-Users] Broken MOH   Andrew Gough
• [Asterisk-Users] RE: Rolling dialplan... best practice?   Ryan Booz
• [Asterisk-Users] Queues and Agents   Nicolás Gudiño
• [Asterisk-Users] Polycom 500 IP and problems with show hints - solved   Kib Eki
• [Asterisk-Users] SIP Subscriptions   Alex Vishnev
• [Asterisk-Users] Asterisk server to provide virtuals IPBX   Christopher L. Wade
• [Asterisk-Users] Queues and Agents   Johann
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] RE: SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] Asterisk server to provide virtuals IPBX   Christopher L. Wade
• [Asterisk-Users] SIP Subscriptions   Douglas Garstang
• [Asterisk-Users] Identifying Frame Slips from PRI debug   ewr at erols.com
• [Asterisk-Users] Unicall E1 Error in Mexico   Steve Underwood
• [Asterisk-Users] turn off message "Silence suppression ..." on Asterisk console   Olle E Johansson
• [Asterisk-Users] Instalar Ubuntu   Moises Silva
• [Asterisk-Users] Asterisk server to provide virtuals IPBX   Olle E Johansson
• [Asterisk-Users] recieve mutiple inbound calls   Andrew Gough
• [Asterisk-Users] Re: Re: RFC 3262 PRACK (Olle E. Johansson)   Olle E Johansson
• [Asterisk-Users] Need help with script from http://www.voip-info.org/wiki/view/Polycom+auto-answer+config   Kib Eki
• [Asterisk-Users] Asterisk server to provide virtuals IPBX   C F
• [Asterisk-Users] SIP Subscriptions   Olle E Johansson
• [Asterisk-Users] SIP Subscriptions   Olle E Johansson
• [Asterisk-Users] SIP Subscriptions   Jason Becker
• [Asterisk-Users] Some values ignored when using static realtime   Simone Ricci
• [Asterisk-Users] SIP Subscriptions   Vladimir Montealegre
• [Asterisk-Users] realtime sip firends not being updated   Tim Connolly
• [Asterisk-Users] Queues and Agents   Johann
• [Asterisk-Users] Queues and Agents   Nicolás Gudiño
• [Asterisk-Users] Name file automatic   asterisk183
• [Asterisk-Users] Crash   Philip Meier
• [Asterisk-Users] no subject   Philip Meier
• [Asterisk-Users] Caller ID   ahmed kassim
• [Asterisk-Users] Tracing a crash with CAPI calls   Andrew Gough
• [Asterisk-Users] Caller ID   Mojo with Horan & Company, LLC
• [Asterisk-Users] Aastra 9133i directory list downloading   Robert La Ferla
• [Asterisk-Users] realtime sip firends not being updated   Aaron Daniel
• [Asterisk-Users] Asterisk's VoiceMail server accessed through various DIDs   ammad jami
• [Asterisk-Users] enabling while/endwhile   jeff oconnell
• [Asterisk-Users] Instalar Ubuntu   Ross C
• [Asterisk-Users] Identifying Frame Slips from PRI debug   tim panton
• [Asterisk-Users] chan_capi-cm 0.6.1 won't load   Johan Helsingius
• [Asterisk-Users] MWI not working - using seperate vm and call routers:   Tim Connolly
• [Asterisk-Users] New To Asterisk/POTS - Hardware Setup Question   john at cusick.ws
• [Asterisk-Users] Extension cannot match ! receiving call mISDN ...   Joao Correia
• [Asterisk-Users] New To Asterisk/POTS - Hardware Setup Questi on   Colin Anderson
• [Asterisk-Users] New To Asterisk/POTS - Hardware Setup Questi on   john at cusick.ws
• [Asterisk-Users] New To Asterisk/POTS - Hardware Setup Questi on   John Millican
• [Asterisk-Users] Asterisk/Zaptel on Kernel 2.6 and ACPI   Waldo Rubinstein
• [Asterisk-Users] FAX Problems - PRI, Adtran and ZetaFax   Jesus Mogollon
• [Asterisk-Users] SIP Subscriptions   Rob Lith
• [Asterisk-Users] MWI not working - using seperate vm and call routers:   Olle E Johansson
• [Asterisk-Users] MWI not working - using seperate vm and call routers:   Aaron Daniel
• [Asterisk-Users] DaemonTools Supervise   Peder at NetworkOblivion
• [Asterisk-Users] Identifying Frame Slips from PRI debug   ewr at erols.com
• [Asterisk-Users] FAX Problems - PRI, Adtran and ZetaFax   Colin Anderson
• [Asterisk-Users] MWI not working - using seperate vm and callrouters:   Tim Connolly
• [Asterisk-Users] IAXy provisioning not working   Leandro Martini - ISAT DGL
• [Asterisk-Users] caller_id and law   Innocent Evil
• [Asterisk-Users] caller_id and law   trixter aka Bret McDanel
• [Asterisk-Users] caller_id and law   Innocent Evil
• [Asterisk-Users] SIP configuration for uip200   Steven Job
• [Asterisk-Users] caller_id and law   Rusty Dekema
• [Asterisk-Users] Unicall E1 Error in Mexico   Jorge Cisneros
• [Asterisk-Users] caller_id and law   Paul
• [Asterisk-Users] caller_id and law   Innocent Evil
• [Asterisk-Users] Help Debugging Dropped Call Audio - Add'l Info   Matt Roth
• [Asterisk-Users] IAXy provisioning not working   Mojo with Horan & Company, LLC
• [Asterisk-Users] inbound routing with amp and TDM400   Tom Rymes
• [Asterisk-Users] RE: Can't pass variables using Originate in PHPAGI 2.14   Anish Basu
• [Asterisk-Users] Help Debugging Dropped Call Audio - Add'l Info   tim panton
• [Asterisk-Users] Can't pass variables using Originate in PHPAGI 2.14   Anish Basu
• [Asterisk-Users] (no subject)   abhishek
• [Asterisk-Users] 3 Phone Call Qualtiy Issues   Jerry Jones
• [Asterisk-Users] recieve mutiple inbound calls   pdhales at optusnet.com.au
• [Asterisk-Users] Identifying Frame Slips from PRI debug   Jerry Jones
• [Asterisk-Users] Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!   Cory Andrews
• [Asterisk-Users] Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!   Jean-Rene Cormier
• [Asterisk-Users] Weird rtpmap issue   Adrian A
• [Asterisk-Users] Calls not incoming to any extension from remote proxy server   abhishek
• [Asterisk-Users] dtmf problem   Bartosz Wegrzyn - asterisk
• [Asterisk-Users] Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!   Tony Vickers
• [Asterisk-Users] Daily Phreak - Daily Telecom, Asterisk and Phreaking Updates   asterisk at dailyphreak.com
• [Asterisk-Users] Daily Phreak - Daily Telecom, Asterisk and Phreaking Updates   Jason Brashear
• [Asterisk-Users] How to record a call   Asterisk Mail
• [Asterisk-Users] Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!   tracinet
• [Asterisk-Users] Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!   Kevin P. Fleming
• [Asterisk-Users] Distinctive Ring and zapata.conf   Robert La Ferla
• [Asterisk-Users] How to record a call   Stefan Reuter
• [Asterisk-Users] Identifying Frame Slips from PRI debug   ewr at erols.com
• [Asterisk-Users] MWI not working - using seperate vm and callrouters:   Olle E Johansson
• [Asterisk-Users] Asterisk server to provide virtuals IPBX   Olle E Johansson
• [Asterisk-Users] Tracing a crash with CAPI calls   Armin Schindler
• [Asterisk-Users] chan_capi-cm 0.6.1 won't load   Armin Schindler
• [Asterisk-Users] Calls not incoming to any extension from remoteproxy server   bbench at mail.bg
• [Asterisk-Users] How to record a call   Dmitry Ivanov
• [Asterisk-Users] Problem with octobri and x100p clone   Agustin Gudiño
• [Asterisk-Users] Problem with octobri and x100p clone   DRi at b-w-computer.de
• [Asterisk-Users] Re: Postgres   Tomislav Parcina
• [Asterisk-Users] Re: Asterisk Call Forwarding   Tomislav Parcina
• [Asterisk-Users] Asterisk server to provide virtuals IPBX   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Re: Aastra 9133i directory list downloading   Tomislav Parcina
• [Asterisk-Users] php agi problem (perhaps problem..)   asterisk at frameweb.it
• [Asterisk-Users] Asterisk server to provide virtuals IPBX   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Problem with octobri and x100p clone   Agustin Gudiño
• [Asterisk-Users] chan_capi-cm 0.6.1 won't load   Johan Helsingius
• [Asterisk-Users] unplugging E1 cables while asterisk running   Simone Cittadini
• [Asterisk-Users] realtime & SIP   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Asterisk <-> Gizmo   Leif Neland
• [Asterisk-Users] IAX No Authority found   Leandro Martini - ISAT DGL
• [Asterisk-Users] *1.2.1 setcidnum from Zap   Vit Bohacek
• [Asterisk-Users] Re: Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!   Tomislav Parcina
• [Asterisk-Users] Re: How to record a call   Tomislav Parcina
• [Asterisk-Users] Problem with octobri and x100p clone   Tzafrir Cohen
• [Asterisk-Users] Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!   Tzafrir Cohen
• [Asterisk-Users] Problem with octobri and x100p clone   Remco Barende
• [Asterisk-Users] PRI problems: B-Channel restart   Antoine Megalla
• [Asterisk-Users] Codec selection in dialplan   Dmitry Ivanov
• [Asterisk-Users] Re: How to record a call   Francesco Peeters (Asterisk)
• [Asterisk-Users] Problem with octobri and x100p clone   Tzafrir Cohen
• [Asterisk-Users] snom Firmware 5.0.   Usman Tahir
• [Asterisk-Users] Anyone doing NAT through m0n0Wall?   Mark Phillips
• [Asterisk-Users] DTMF - > FSK CallerID problems   Ossi Sariola
• [Asterisk-Users] IAX No Authority found   Mike Safford
• [Asterisk-Users] need help in building dynamic conference   vicky sarathy
• [Asterisk-Users] IAX No Authority found   bbench at mail.bg
• [Asterisk-Users] agentcallbacklogin   Dov Bigio
• [Asterisk-Users] Re: Re: How to record a call   Tomislav Parcina
• [Asterisk-Users] unplugging E1 cables while asterisk running   Calin Serbanescu
• [Asterisk-Users] anybody getting "No authority found" with teliax now?   Thomas Miller
• [Asterisk-Users] anybody getting "No authority found" with teliaxnow?   Jonathan k. Creasy
• [Asterisk-Users] PRI problems: B-Channel restart   Michael Sampson
• [Asterisk-Users] Re: Re: How to record a call   Blake Krone
• [Asterisk-Users] RE: Can't pass variables using Originate in PHPAGI 2.14   Matt Riddell
• [Asterisk-Users] chan_oss.so   Tomislav Parcina
• [Asterisk-Users] RE: Can't pass variables using Originate in PHPAGI 2.14   Anish Basu
• [Asterisk-Users] Re: Re: How to record a call   Francesco Peeters (Asterisk)
• [Asterisk-Users] Weird Echo Problem   bails
• [Asterisk-Users] asterisk AVM C2 again   stéphane plichon
• [Asterisk-Users] Anyone doing NAT through m0n0Wall?   Colin Anderson
• [Asterisk-Users] Weird Echo Problem   Kevin P. Fleming
• [Asterisk-Users] Anyone doing NAT through m0n0Wall?   Francis Ballares (VoIPware.ca)
• [Asterisk-Users] unplugging E1 cables while asterisk running   C F
• [Asterisk-Users] chan_oss.so   Saul Diaz
• [Asterisk-Users] chan_oss.so   Doug Lytle
• [Asterisk-Users] TDM2400   Guillermo Salas M
• [Asterisk-Users] TDM2400   BJ Weschke
• [Asterisk-Users] Anyone doing NAT through m0n0Wall?   Mark Phillips
• [Asterisk-Users] agentcallbacklogin   BJ Weschke
• [Asterisk-Users] Weird Echo Problem   Nathan C. Smith
• [Asterisk-Users] recording queue calls   Dov Bigio
• [Asterisk-Users] Realtime Config   Douglas Garstang
• [Asterisk-Users] 1.2.0 PRI dropping calls occasionally...   David C. Nicosia
• [Asterisk-Users] RE: Realtime Config   Douglas Garstang
• [Asterisk-Users] Realtime Config   Aaron Daniel
• [Asterisk-Users] SIP - SIP bridge dropping calls?   David C. Nicosia
• [Asterisk-Users] Realtime Config   Douglas Garstang
• [Asterisk-Users] Zap Error   Goran Donev
• [Asterisk-Users] TDM2400   Massimo De Nadal
• [Asterisk-Users] Make Asterisk explicitly UNREGISTER from a SIP service?   Tony Mountifield
• [Asterisk-Users] Manager API connections - crashes?   rushowr
• [Asterisk-Users] how to follow a call in the console   Patrick Fortin
• [Asterisk-Users] asterisk AVM C2 again   Armin Schindler
• [POSSIBLE SPAM] RE: [Asterisk-Users] Identifying Frame Slips from PRI debug   ewr at erols.com
• [Asterisk-Users] Re: how to follow a call in the console   Steven
• [Asterisk-Users] Asterfax beta4, Asterisk 1.2.0 and issue sending FAX   gianrico at gianrico.com
• [Asterisk-Users] CVS problem?   Colin Anderson
• [Asterisk-Users] CVS problem?   Darren Wiebe
• [Asterisk-Users] CVS problem?   Colin Anderson
• [Asterisk-Users] wav to g729   Innocent Evil
• [Asterisk-Users] CVS problem?   Kevin P. Fleming
• [Asterisk-Users] CVS problem?   Colin Anderson
• [Asterisk-Users] anybody getting "No authority found" with teliaxnow?   Jimmy Smith
• [Asterisk-Users] wav to g729   Matt Riddell
• [Asterisk-Users] CVS problem?   Darren Wiebe
• [Asterisk-Users] Re: Teliax billing question   Jimmy Smith
• [Asterisk-Users] CVS problem?   Kevin P. Fleming
• [Asterisk-Users] TDM2400   Vahan Yerkanian
• [Asterisk-Users] wav to g729   Innocent Evil
• [Asterisk-Users] ast_sock_cmd: pipe commands to asterisk   Tzafrir Cohen
• [Asterisk-Users] CVS problem?   Colin Anderson
• [Asterisk-Users] Asterisk server to provide virtuals IPBX   Kevin Kiely
• [Asterisk-Users] Asterisk server to provide virtuals IPBX   C F
• [Asterisk-Users] App_valetparking   Kevin Kiely
• [Asterisk-Users] wav to g729   listas iPfone
• [Asterisk-Users] sill looking for a provider   Jimmy Smith
• [Asterisk-Users] chan_sip.c error message   Ivan Lopez
• [Asterisk-Users] forwarding a caller to a conference room   Asterisk Mail
• [Asterisk-Users] wav to g729   Innocent Evil
• [Asterisk-Users] Fwd: Legacy PBX -> * -> Voip Calls problems   bbench at mail.bg
• [Asterisk-Users] wav to g729   Kristian Kielhofner
• [Asterisk-Users] "CALLED NUMBER" in IAX2   Michael Stearne
• [Asterisk-Users] unplugging E1 cables while asterisk running   Simone Cittadini
• [Asterisk-Users] wav to g729   Innocent Evil
• [Asterisk-Users] Help Debugging Dropped Call Audio - Add'l Info   Martin Joseph
• [Asterisk-Users] Re: Weird rtpmap issue   Adrian A
• [Asterisk-Users] Calls not incoming to any extension from remoteproxy server   abhishek
• [Asterisk-Users] wav to g729   Innocent Evil
• [Asterisk-Users] wav to g729   Kristian Kielhofner
• [Asterisk-Users] ast_sock_cmd: pipe commands to asterisk   Tzafrir Cohen
• [Asterisk-Users] anybody getting "No authority found" with teliax now?   Martin Joseph
• [Asterisk-Users] wav to g729   John Reynolds
• [Asterisk-Users] "CALLED NUMBER" in IAX2   Rehan Ahmed
• [Asterisk-Users] SNOM 360 locked up   Steven Ringwald
• [Asterisk-Users] wav to g729   Kristian Kielhofner
• [Asterisk-Users] Help Debugging Dropped Call Audio - Add'l Info   Matt Roth
• [Asterisk-Users] Re: Fw: Legacy PBX -> * -> Voip Calls problems   bbench at mail.bg
• [Asterisk-Users] Creating conf files from db   Douglas Garstang
• [Asterisk-Users] Asterisk server to provide virtuals IPBX   Rehan Ahmed
• [Asterisk-Users] Creating conf files from db   Jean-Michel Hiver
• [Asterisk-Users] Creating conf files from db   Douglas Garstang
• [Asterisk-Users] SNOM 360 locked up   Christian Stredicke
• [Asterisk-Users] Creating conf files from db   Azfhasterisk
• [Asterisk-Users] Creating conf files from db   Douglas Garstang
• [Asterisk-Users] Creating conf files from db   Ariel Batista
• [Asterisk-Users] FOLLOWUP: .call files on PRI / Zombie AGI proces ses in FC2 / 1.2 Beta 1 under load   Colin Anderson
• [Asterisk-Users] Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!   Bryan J. Smith
• [Asterisk-Users] SIP - SIP bridge dropping calls?   Adrian A
• [Asterisk-Users] Identifying Frame Slips from PRI debug   ewr at erols.com
• [Asterisk-Users] Asterisk@Home Fax to Email problems   Jay Nemrow
• [Asterisk-Users] Tracing a crash with CAPI calls   Andrew Gough
• [Asterisk-Users] Asterisk@Home Fax to Email problems   trixter aka Bret McDanel
• [Asterisk-Users] Asterisk@Home Fax to Email problems   Jorge Cisneros
• [Asterisk-Users] SNOM 360 locked up   Steven Ringwald
• [Asterisk-Users] SNOM 360 locked up SOLVED   Steven Ringwald
• [Asterisk-Users] automon doesn't work with 1.0.9   Asterisk Mail
• [Asterisk-Users] Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!   David Muench
• [Asterisk-Users] automon doesn't work with 1.0.9   Kevin P. Fleming
• [Asterisk-Users] Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!   Kevin P. Fleming
• [Asterisk-Users] Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!   Master Abi
• [Asterisk-Users] What hardware fits my needs?   S.Ammad Jami
• [Asterisk-Users] need help in building dynamic conference   vicky sarathy
• [Asterisk-Users] Asterisk@Home Fax to Email problems   Technical Support
• [Asterisk-Users] Emergency Information Needed: sip.conf - bindport allow multiple ports?   rushowr
• [Asterisk-Users] Emergency Information Needed: sip.conf - bindport allow multiple ports?   BJ Weschke
• [Asterisk-Users] ast_sock_cmd: pipe commands to asterisk   Dinesh Nair
• [Asterisk-Users] Emergency Information Needed: sip.conf -bindport allow multiple ports?   rushowr
• [Asterisk-Users] Malformed CallerID freaks out SIP channel   Brian Capouch
• [Asterisk-Users] Emergency Information Needed: sip.conf -bindport allow multiple ports?   BJ Weschke
• [Asterisk-Users] Malformed CallerID freaks out SIP channel   BJ Weschke
• [Asterisk-Users] Emergency Information Needed: sip.conf -bindportallow multiple ports?   rushowr
• [Asterisk-Users] Creating conf files from db   Peter Bowyer
• [Asterisk-Users] No sound problem, chan_sip.c:3451   Jan du Toit
• [Asterisk-Users] dialing outbound using te411p   Derek O'Flynn
• [Asterisk-Users] Malformed CallerID freaks out SIP channel   Brian Capouch
• [Asterisk-Users] Grandstream Budge Tone 102   Tomislav Parcina
• [Asterisk-Users] Problems with current chan-capi-cm [SOLVED]   Peer Oliver Schmidt
• [Asterisk-Users] asterisk AVM C2 again   stéphane plichon
• [Asterisk-Users] Merry Xmas to everybody!   Mauro Zanin
• [Asterisk-Users] Feature: Attendet transfer with original caller ID   Kib Eki
• [Asterisk-Users] no have dial tone   alfa
• [Asterisk-Users] Virtual Memory Usage   Christian Benke
• [Asterisk-Users] Merry Xmas to everybody!   Francesco Peeters (Asterisk)
• [Asterisk-Users] chinese asterisk related web site opened......   Jeffery Chen
• [Asterisk-Users] Merry Xmas to everybody!   Dmitry Ivanov
• [Asterisk-Users] Strange Problem   Gulzar Hussain
• [Asterisk-Users] Is there a GUI for asterisk realtime   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Re: chinese asterisk related web site opened......   Jeffery Chen
• [Asterisk-Users] Re: chinese asterisk related web site opened......   Jeffery Chen
• [Asterisk-Users] Tracing a crash with CAPI calls   Armin Schindler
• [Asterisk-Users] Problems with current chan-capi-cm [SOLVED]   Armin Schindler
• [Asterisk-Users] asterisk AVM C2 again   Armin Schindler
• [Asterisk-Users] What hardware fits my needs?   asterisk-users at ecessor.com
• [Asterisk-Users] Transfer   Tomislav Parcina
• [Asterisk-Users] Re: Re: Re: How to record a call   Tomislav Parcina
• [Asterisk-Users] Will ooh323 ever move from addons?   Dinesh Nair
• [Asterisk-Users] Transfer   Jean-Michel Hiver
• [Asterisk-Users] ast_sock_cmd: pipe commands to asterisk   Tzafrir Cohen
• [Asterisk-Users] Anyone doing NAT through m0n0Wall?   Dinesh Nair
• [Asterisk-Users] Grandstream Budge Tone 102   Mark Phillips
• [Asterisk-Users] ast_sock_cmd: pipe commands to asterisk   Dinesh Nair
• [Asterisk-Users] What hardware fits my needs?   Mark Phillips
• [Asterisk-Users] asterisk AVM C2 again   stéphane plichon
• [Asterisk-Users] Virtual Memory Usage   Tzafrir Cohen
• [Asterisk-Users] Grandstream Budge Tone 102   Bob Goddard
• [Asterisk-Users] Creating conf files from db   Tzafrir Cohen
• [Asterisk-Users] Emergency Information Needed: sip.conf -bindport allow multiple ports?   Rich Adamson
• [Asterisk-Users] Creating conf files from db   Jithendra
• [Asterisk-Users] caller id   Dov Bigio
• [Asterisk-Users] Virtual Memory Usage   Rich Adamson
• [Asterisk-Users] How to make Asterisk to generate and terminate calls   Ravi Shankar
• [Asterisk-Users] No sound problem, chan_sip.c:3451   Kevin P. Fleming
• [Asterisk-Users] Will ooh323 ever move from addons?   Kevin P. Fleming
• [Asterisk-Users] how to get the number of an external phone with Asterisk Manager   amaury BOSSE
• [Asterisk-Users] Re: chan_oss.so   Tomislav Parcina
• [Asterisk-Users] How to make Asterisk to generate and terminatecalls   Shawn Porter
• [Asterisk-Users] How to make Asterisk to generate and terminate calls   Mark Phillips
• [Asterisk-Users] Grandstream Budge Tone 102   Mark Phillips
• [Asterisk-Users] no have dial tone   Stewart Nelson
• [Asterisk-Users] Re: Grandstream Budge Tone 102   Tomislav Parcina
• [Asterisk-Users] Re: Grandstream Budge Tone 102   Tomislav Parcina
• [Asterisk-Users] Transfer   Henri Herscher
• [Asterisk-Users] Transfer   Philipp von Klitzing
• [Asterisk-Users] Does broadvoice modify caller ID name?   Paul
• [Asterisk-Users] Creating conf files from db   Douglas Garstang
• [Asterisk-Users] Asterisk@Home Fax to Email problems   Michael Sampson
• [Asterisk-Users] Virtual Memory Usage   Bob Goddard
• [Asterisk-Users] Re: Grandstream Budge Tone 102   Bob Goddard
• [Asterisk-Users] Manager API connections - crashes?   Moises Silva
• [Asterisk-Users] FYI on zttool output on SMP system   Steven
• [Asterisk-Users] DaemonTools Supervise   Moises Silva
• [Asterisk-Users] wav to g729   Max Blackmer
• [Asterisk-Users] What hardware fits my needs?   Michael Sampson
• [Asterisk-Users] provu 2100 videophone and asterisk   gianrico at gianrico.com
• [Asterisk-Users] SIP permit/deny   Douglas Garstang
• [Asterisk-Users] Creating conf files from db   Peter Bowyer
• [Asterisk-Users] What hardware fits my needs?   S.Ammad Jami
• [Asterisk-Users] SIP permit/deny   Aaron Daniel
• [Asterisk-Users] Anyone doing NAT through m0n0Wall?   Manny A. Wise
• [Asterisk-Users] Anyone doing NAT through m0n0Wall?   Manny A. Wise
• [Asterisk-Users] SIP permit/deny   Douglas Garstang
• [Asterisk-Users] Does broadvoice modify caller ID name?   Mark Phillips
• [Asterisk-Users] Merry Xmas to everybody!   Mark Phillips
• [Asterisk-Users] List Of Defined Variables   Michael Stearne
• [Asterisk-Users] RE: SIP permit/deny   Douglas Garstang
• [Asterisk-Users] Merry Xmas to everybody!   FaberK
• [Asterisk-Users] Re: List Of Defined Variables   C F
• [Asterisk-Users] RE: SIP permit/deny   Aaron Daniel
• [Asterisk-Users] wav to g729   Carlos Rojas
• [Asterisk-Users] RE: SIP permit/deny   Douglas Garstang
• [Asterisk-Users] Does broadvoice modify caller ID name?   Kevin P. Fleming
• [Asterisk-Users] RE: SIP permit/deny   Kevin P. Fleming
• [Asterisk-Users] Does broadvoice modify caller ID name?   Paul
• [Asterisk-Users] RE: SIP permit/deny   Douglas Garstang
• [Asterisk-Users] Re: List Of Defined Variables   Michael Stearne
• [Asterisk-Users] Re: List Of Defined Variables   Michael Stearne
• [Asterisk-Users] Matching SIP users and peers   Douglas Garstang
• [Asterisk-Users] Matching SIP users and peers   Joshua Colp
• [Asterisk-Users] Re: List Of Defined Variables   Johann
• [Asterisk-Users] Matching SIP users and peers   BJ Weschke
• [Asterisk-Users] Matching SIP users and peers   C F
• [Asterisk-Users] RE: SIP permit/deny   Kevin P. Fleming
• [Asterisk-Users] Matching SIP users and peers   Douglas Garstang
• [Asterisk-Users] Matching SIP users and peers   Kevin P. Fleming
• [Asterisk-Users] Is there a GUI for asterisk realtime   Darren Wiebe
• [Asterisk-Users] Matching SIP users and peers   Douglas Garstang
• [Asterisk-Users] Matching SIP users and peers   C F
• [Asterisk-Users] TDM2400P driver change   Malcolm Davenport
• [Asterisk-Users] Matching SIP users and peers   Kevin P. Fleming
• [Asterisk-Users] TDM2400P driver change   Kerry Garrison
• [Asterisk-Users] RE: IAX Call Pickup   Steve Gladden
• [Asterisk-Users] TDM2400P driver change   Rich Adamson
• [Asterisk-Users] Asterisk Configuration   Faheem Ahmed
• [Asterisk-Users] Is there a GUI for asterisk realtime   Paul Mahler
• [Asterisk-Users] Asterisk Configuration   Jonathan Augenstine
• [Asterisk-Users] Probs with outbound calls   Colin Tordale
• [Asterisk-Users] ASterisk and home lines.. DGM-TDM01B or x100 ?   Jimmy Smith
• [Asterisk-Users] tdm400 fxo problem   Filippo Carone
• [Asterisk-Users] AMP stuff via CLI?   Ken D'Ambrosio
• [Asterisk-Users] problem with tdm400 fxo   Filippo Carone
• [Asterisk-Users] tdm400 fxo problem   Kerry Garrison
• [Asterisk-Users] ASterisk and home lines.. DGM-TDM01B or x100 ?   Ben Higley
• [Asterisk-Users] ASterisk and home lines.. DGM-TDM01B or x100 ?   Jimmy Smith
• [Asterisk-Users] Re: List Of Defined Variables   Michael Stearne
• [Asterisk-Users] tdm400 fxo problem   Rich Adamson
• [Asterisk-Users] ASterisk and home lines.. DGM-TDM01B or x100 ?   Ben Higley
• [Asterisk-Users] Aastra firmware 1.3.x (Far-End sound level issue)   Taco Scargo
• [Asterisk-Users] ASterisk and home lines.. DGM-TDM01B or x100 ?   Jimmy Smith
• [Asterisk-Users] Merry Xmas to everybody!   tracinet
• [Asterisk-Users] Merry Xmas to everybody!   Guillermo Salas M
• [Asterisk-Users] tdm400 fxo problem   James B. MacLean
• [Asterisk-Users] Strange Problem   Rehan Ahmed
• [Asterisk-Users] Asterisk With Yahoo messenger   Rehan Ahmed
• [Asterisk-Users] Aastra firmware 1.3.x (Far-End sound level issue)   Robert La Ferla
• [Asterisk-Users] OT: Please recommend VoIP service providers for US to Japan calls   Robert La Ferla
• [Asterisk-Users] Problem with Xlite free phone(Xten)   abhishek
• [Asterisk-Users] Feature: Attendet transfer with original caller ID   Alexander Lopez
• [Asterisk-Users] How to make Asterisk to generate and terminate calls   Ravi Shankar
• [Asterisk-Users] Will ooh323 ever move from addons?   Dinesh Nair
• [Asterisk-Users] Callerid   Code Lover
• [Asterisk-Users] Aastra firmware 1.3.x (Far-End sound level issue)   Taco Scargo
• [Asterisk-Users] New To Asterisk/POTS - Hardware Setup Question   Martin Joseph
• [Asterisk-Users] Matching SIP users and peers   Luigi Rizzo
• [Asterisk-Users] tdm400 fxo problem   Filippo Carone
• [Asterisk-Users] tdm400 fxo problem   Filippo Carone
• [Asterisk-Users] tdm400 fxo problem   Filippo Carone
• [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with ringing   Michael J. Tubby G8TIC
• [Asterisk-Users] Laptop PCMCIA ISDN card   Sascha Andres
• [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with ringing   Armin Schindler
• [Asterisk-Users] CAPI and *   Sascha Andres
• [Asterisk-Users] CAPI and *   Armin Schindler
• [Asterisk-Users] Best Voip provider   jonny hashem
• [Asterisk-Users] Broken sound Music on hold, , voice prompts good   Zeeshan
• [Asterisk-Users] AMP stuff via CLI?   Tzafrir Cohen
• [Asterisk-Users] Best Voip provider   trixter aka Bret McDanel
• [Asterisk-Users] Dialling out with clone X100P board   Roger Hill
• [Asterisk-Users] Dialling out with clone X100P board   burke at tailorhosting.com
• [Asterisk-Users] Best Voip provider   chawki hammoud
• [Asterisk-Users] Best Voip provider   trixter aka Bret McDanel
• [Asterisk-Users] recording queue calls   Faris Raouf
• [Asterisk-Users] recording queue calls   Tom Lynn
• [Asterisk-Users] in and out recorded audio mixing in queues   Faris Raouf
• [Asterisk-Users] recording queue calls   Faris Raouf
• [Asterisk-Users] PRI outgoing caller ID stopped working   Andrew Kohlsmith
• [Asterisk-Users] recording queue calls   Tom Lynn
• [Asterisk-Users] Broken sound Music on hold, , voice prompts good   Mark Phillips
• [Asterisk-Users] Callerid   Mark Phillips
• [Asterisk-Users] Broken sound Music on hold, , voice prompts good   Zeeshan
• [Asterisk-Users] Broken sound Music on hold, , voice prompts good   Luki
• [Asterisk-Users] PRI outgoing caller ID stopped working   Kevin P. Fleming
• [Asterisk-Users] System(...) but how to pass parameters?   Pisac
• [Asterisk-Users] System(...) but how to pass parameters?   Pisac
• [Asterisk-Users] Merry Xmas to everybody!   Carlos Rojas
• [Asterisk-Users] System(...) but how to pass parameters?   trixter aka Bret McDanel
• [Asterisk-Users] System(...) but how to pass parameters?   trixter aka Bret McDanel
• [Asterisk-Users] Latest Source   Carlos Rojas
• [Asterisk-Users] Busy signal for incoming calls from broadvoice   Robert La Ferla
• [Asterisk-Users] Busy signal for incoming calls from broadvoice   trixter aka Bret McDanel
• [Asterisk-Users] Busy signal for incoming calls from broadvoice   Paul
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 148   alum
• [Asterisk-Users] Busy signal for incoming calls from broadvoice   Robert La Ferla
• [Asterisk-Users] Callerid   Abdul Lateef
• [Asterisk-Users] Best Voip provider   Abdul Lateef
• [Asterisk-Users] Virtual Memory Usage   Tzafrir Cohen
• [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with ringing   Michael J. Tubby G8TIC
• [Asterisk-Users] Busy signal for incoming calls from broadvoice   asterisk-users at ecessor.com
• [Asterisk-Users] TE410P E1 Red Alarm   Diyanat Ali
• [Asterisk-Users] TE410P E1 Red Alarm   Mark Edwards
• [Asterisk-Users] TE410P E1 Red Alarm   Rich Adamson
• [Asterisk-Users] How to make Asterisk to generate and terminate calls   Ravi Shankar
• [Asterisk-Users] WMI Problem   Il Neofita
• [Asterisk-Users] help   Leonardo Alves
• [Asterisk-Users] help   C F
• [Asterisk-Users] help   trixter aka Bret McDanel
• [Asterisk-Users] s or _X. , is there any change since Asterisk 1.2   Zeeshan
• [Asterisk-Users] Busy signal for incoming calls from broadvoice   Robert La Ferla
• [Asterisk-Users] newbie question about making outbound call   Jason D. Wolfe
• [Asterisk-Users] help   Dualcall.com
• [Asterisk-Users] Cisco PGW-2200 OR Asterisk   Abdul Lateef
• [Asterisk-Users] Affordable IP Phones for Asterisk   Sam Tam
• [Asterisk-Users] weird problem with sipura spa2000 and sound card pa setup   lee at michwave.com
• [Asterisk-Users] weird problem with sipura spa2000 and sound cardpa setup   Alexander Lopez
• [Asterisk-Users] weird problem with sipura spa2000 and sound cardpa setup   lee at michwave.com
• [Asterisk-Users] weird problem with sipura spa2000 and soundcardpa setup   Alexander Lopez
• [Asterisk-Users] Channel bank timing   Chris Mason (Lists)
• [Asterisk-Users] Channel bank timing   Rich Adamson
• [Asterisk-Users] Channel bank timing   Andrew Kohlsmith
• [Asterisk-Users] Channel bank timing   Andrew Kohlsmith
• [Asterisk-Users] Grandstream Budge Tone 102   Saul Diaz
• [Asterisk-Users] Channel bank timing   Chris Mason
• [Asterisk-Users] Channel bank timing   Andrew Kohlsmith
• [Asterisk-Users] Problem with date & time on Aastra 480i since release 1.3   Jacques Leisy
• [Asterisk-Users] Why no sound from festival?   Robert La Ferla
• [Asterisk-Users] Problem with date & time on Aastra 480i since release 1.3   Robert La Ferla
• [Asterisk-Users] weird problem with sipura spa2000 and soundcardpa setup   lee at michwave.com
• [Asterisk-Users] No of records in calls table   Kanishka Somaratne
• [Asterisk-Users] Asterisk Christmas Help request   Mr Asterisk
• [Asterisk-Users] Asterisk Christmas Help request   kevin ling
• [Asterisk-Users] NEW Asterisk Management Interface with Java Manager Live Console.   Vikram Rangnekar
• [Asterisk-Users] Aastra firmware 1.3.x (Far-End sound level issue)   BennyBad
• [Asterisk-Users] Channel bank timing   Chris Mason (Lists)
• [Asterisk-Users] Channel bank timing   Rich Adamson
• [Asterisk-Users] Channel bank timing   Andrew Kohlsmith
• [Asterisk-Users] Eicon DIVA Server V-BRI questions   Jens Vagelpohl
• [Asterisk-Users] Creating conf files from db   Jithendra
• [Asterisk-Users] RE: how to make contribution in asterisk   Tejas Shah
• [Asterisk-Users] Eicon DIVA Server V-BRI questions   Jens Vagelpohl
• [Asterisk-Users] problem with tdm400 fxo   Aryanto Rachmad
• [Asterisk-Users] Re: Eicon DIVA Server V-BRI questions   Stefan Tichy
• [Asterisk-Users] Re: Eicon DIVA Server V-BRI questions   Jens Vagelpohl
• [Asterisk-Users] NEW Asterisk Management Interface with JavaManager Live Console.   Steve Totaro
• [Asterisk-Users] Asterisk Christmas Help request   Doug Lytle
• [Asterisk-Users] Asterisk Christmas Help request   Doug Lytle
• [Asterisk-Users] channel monitoring whisper mode?   Script Head
• [Asterisk-Users] channel monitoring whisper mode?   C F
• [Asterisk-Users] Problem with date & time on Aastra 480i since release 1.3   Jacques Leisy
• [Asterisk-Users] weird problem with sipura spa2000 and soundcardpa setup   Jorge Cisneros
• [Asterisk-Users] Problem with date & time on Aastra 480i since release 1.3   Robert La Ferla
• [Asterisk-Users] RE: how to make contribution in asterisk   Simone Cittadini
• [Asterisk-Users] Re: Eicon DIVA Server V-BRI questions   Stefan Tichy
• [Asterisk-Users] NEW Asterisk Management Interface withJavaManager Live Console.   Quark IT - Hilton Travis
• [Asterisk-Users] Delays in IVR   Eric "ManxPower" Wieling
• [Asterisk-Users] Delays in IVR   Adam Moffett
• [Asterisk-Users] Delays in IVR   BJ Weschke
• [Asterisk-Users] Asterisk lines go into PBX?   Doug
• [Asterisk-Users] 64 bit Zaptel?   Kristian Kielhofner
• [Asterisk-Users] Delays in IVR   Adam Moffett
• [Asterisk-Users] Delays in IVR   Adam Moffett
• [Asterisk-Users] 64 bit Zaptel?   Zoa
• [Asterisk-Users] Asterisk lines go into PBX?   Dean Collins
• [Asterisk-Users] Delays in IVR   Eric "ManxPower" Wieling
• [Asterisk-Users] Delays in IVR   C F
• [Asterisk-Users] Delays in IVR   BJ Weschke
• [Asterisk-Users] Operator breakout from voicemail   Iain Barker
• [Asterisk-Users] weird problem with sipura spa2000 and soundcardpa setup   lee at michwave.com
• [Asterisk-Users] 64 bit Zaptel?   Tony Hoyle
• [Asterisk-Users] Cisco PGW-2200 OR Asterisk   Markku Korpi
• [Asterisk-Users] How to check Digium TE410P firmware version?   Franz Wu
• [Asterisk-Users] Operator breakout from voicemail   Alexander Lopez
• [Asterisk-Users] How to check Digium TE410P firmware version?   BJ Weschke
• [Asterisk-Users] Stay away from Grandstream!   Elene Kinsky
• [Asterisk-Users] Stay away from Grandstream!   Cory Andrews
• [Asterisk-Users] Stay away from Grandstream!   Chris Albertson
• [Asterisk-Users] How to check Digium TE410P firmware version?   Robert La Ferla
• [Asterisk-Users] LD_LIBRARY_PATH   Kanishka Somaratne
• [Asterisk-Users] How to check Digium TE410P firmware version?   BJ Weschke
• [Asterisk-Users] SIP "403 Forbidden" Errors...   Conall O'Brien
• [Asterisk-Users] iptables rules for forwarding SIP/RTP to Asterisk server from behind nat firewall/router   Robert La Ferla
• [Asterisk-Users] Busy signal for incoming calls from broadvoice   Robert La Ferla
• [Asterisk-Users] Re: Busy signal for incoming calls from broadvoice   Tomislav Parcina
• [Asterisk-Users] Pls. explain what happens...   Mauro Zanin
• [Asterisk-Users] Pls. explain what happens...   Alessio Focardi
• [Asterisk-Users] Pls. explain what happens...   Francesco Peeters (Asterisk)
• [Asterisk-Users] iptables rules for forwarding SIP/RTP to Asterisk server from behind nat firewall/router   Aryanto Rachmad
• [Asterisk-Users] Channel bank timing   Dinesh Nair
• [Asterisk-Users] Asterisk+mgcp setup+vrg 121   in out
• [Asterisk-Users] Stay away from Grandstream!   Steve Underwood
• [Asterisk-Users] SIP ENUM Daemon   Nahid Hossain
• [Asterisk-Users] recording queue calls   Dov Bigio
• [Asterisk-Users] Changing Automon filenames?   Francesco Peeters (Asterisk)
• [Asterisk-Users] Stay away from Grandstream!   Rob Hillis
• [Asterisk-Users] Channel bank timing   Andrew Kohlsmith
• [Asterisk-Users] Changing Automon filenames?   BJ Weschke
• [Asterisk-Users] Re: Grandstream Budge Tone 102   Tomislav Parcina
• [Asterisk-Users] Re: Transfer   Tomislav Parcina
• [Asterisk-Users] IAX media path, forcing server to stay in the middle   Simone Cittadini
• [Asterisk-Users] Changing Automon filenames?   Francesco Peeters (Asterisk)
• [Asterisk-Users] Re: Grandstream Budge Tone 102   Michiel van Baak
• [Asterisk-Users] How to check Digium TE410P firmware version?   Kevin P. Fleming
• [Asterisk-Users] Polycom IP301 time changing   Jonathan k. Creasy
• [Asterisk-Users] Delays in IVR   Adam Moffett
• [Asterisk-Users] How to check Digium TE410P firmware version?   Steve Totaro
• [Asterisk-Users] Delays in IVR   BJ Weschke
• [Asterisk-Users] Asterisk on VPS   Ross C
• [Asterisk-Users] IAX media path, forcing server to stay in the middle   Time Bandit
• [Asterisk-Users] Broken sound Music on hold, , voice prompts good   Paul Hewlett
• [Asterisk-Users] spandsp & fax   Dov Bigio
• [Asterisk-Users] Asterisk 1.2.1 and X100 clone Zap problem   Todd Reese
• [Asterisk-Users] IAX media path, forcing server to stay in the middle   Andrew Kohlsmith
• [Asterisk-Users] Re: IAX media path, forcing server to stay in the middle   Simone Cittadini
• [Asterisk-Users] RE: [Asterisk- Pls. explain what happens...   Kevin Steil
• [Asterisk-Users] How to check Digium TE410P firmware version?   Kevin P. Fleming
• [Asterisk-Users] CDR_CSV stops writing, help!   Tyler
• [Asterisk-Users] IAX media path, forcing server to stay in the middle   Kevin P. Fleming
• [Asterisk-Users] Re: Transfer   Tomislav Parcina
• [Asterisk-Users] IAX media path, forcing server to stay in the middle   Time Bandit
• [Asterisk-Users] Re: Transfer   Tomislav Parcina
• [Asterisk-Users] spandsp & fax   BJ Weschke
• [Asterisk-Users] Stay away from Grandstream!   Chris Bagnall
• [Asterisk-Users] spandsp & fax   Kristof Hardy
• [Asterisk-Users] Unicall E1 Error in Mexico   Martinez Felix
• [Asterisk-Users] Extension cannot match ! receiving call mISDN ...   Ivo Simicevic
• [Asterisk-Users] Cisco dtmf   Tomislav Parcina
• [Asterisk-Users] Delays in IVR   Rich Adamson
• [Asterisk-Users] CDR_CSV stops writing, help!   Alexander Lopez
• [Asterisk-Users] TDD/TTY - How does one use this?   Don Fanning
• [Asterisk-Users] "one touch record" on asterisk 1.2.1 uses monitor and not mixmonitor   Vikas
• [Asterisk-Users] newbie question about making outbound call   Moises Silva
• [Asterisk-Users] Callerid ID lookup program updated (CID_rewrite v1.2)   Technical Support
• [Asterisk-Users] spandsp & fax   Dov Bigio
• [Asterisk-Users] "one touch record" on asterisk 1.2.1 uses monitor and not mixmonitor   BJ Weschke
• [Asterisk-Users] Strange IAX messages on the console   Joseph Rothstein
• [Asterisk-Users] Stay away from Grandstream!   The VoIP Connection
• [Asterisk-Users] channel monitoring whisper mode?   Saul Diaz
• [Asterisk-Users] LD_LIBRARY_PATH   Douglas Garstang
• [Asterisk-Users] Login incorrect on ZIP2 phones when checking voicemail   Dan Elder
• [Asterisk-Users] Cisco 7912G through NAT, problems with tones detection.   Diego Mariano Velo
• [Asterisk-Users] Realtime Static/Dynamic Preference   Douglas Garstang
• [Asterisk-Users] Realtime Static/Dynamic Preference   BJ Weschke
• [Asterisk-Users] Polycom IP301 time changing   Mojo with Horan & Company, LLC
• [Asterisk-Users] Login incorrect on ZIP2 phones when checking voicemail   John Novack
• [Asterisk-Users] Realtime Static/Dynamic Preference   Douglas Garstang
• [Asterisk-Users] Blackberry SIM card   Robert Rawlinson
• [Asterisk-Users] Realtime Static/Dynamic Preference   Kevin P. Fleming
• [Asterisk-Users] Blackberry SIM card   Kerry Garrison
• [Asterisk-Users] Transfer   Tobias Wolf
• [Asterisk-Users] asterisk AVM C2 again   Armin Schindler
• [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly   James Sizemore
• [Asterisk-Users] Realtime Static/Dynamic Preference   C F
• [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with ringing   Armin Schindler
• [Asterisk-Users] UK, Disconnect supervision   Peter Hoppe
• [Asterisk-Users] Cisco dtmf   Greg Oliver
• [Asterisk-Users] PRI outgoing caller ID stopped working   Andrew Kohlsmith
• [Asterisk-Users] SIP ENUM Daemon   Klaus Darilion
• [Asterisk-Users] CDR_CSV stops writing, help!   Tyler
• [Asterisk-Users] Asterisk lines go into PBX?   Michael Collins
• [Asterisk-Users] Blackberry SIM card   Alexander Lopez
• [Asterisk-Users] TDM2400   f6hqz-m at hamwlan.net
• [Asterisk-Users] polycom sip slower than grandstream   Dean Collins
• [Asterisk-Users] MSN Messenger / Windows messenger Passport service With asterisk any one ?   Rehan Ahmed
• [Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly   James Sizemore
• [Asterisk-Users] Realtime Static/Dynamic Preference   Douglas Garstang
• [Asterisk-Users] Realtime Static/Dynamic Preference   Douglas Garstang
• [Asterisk-Users] Blackberry SIM card   Kerry Garrison
• [Asterisk-Users] agent logs   Hall, Eric M.
• [Asterisk-Users] Asterisk Realtime Database Redundancy   Douglas Garstang
• [Asterisk-Users] How to check Digium TE410P firmware version?   Franz Wu
• [Asterisk-Users] How to register a sip user/peer in real time   Rehan Ahmed
• [Asterisk-Users] Polycom Soundpoint 501 outbound calls always show NO ANSWER   Reid W. Johnson
• [Asterisk-Users] asterisk AVM C2 again   Dave Cotton
• [Asterisk-Users] Play soundfile before snswer   Arik Funke
• [Asterisk-Users] Play soundfile before snswer   Kerry Garrison
• [Asterisk-Users] asterisk AVM C2 again   Armin Schindler
• [Asterisk-Users] Asterisk on VPS   Blake OPS
• [Asterisk-Users] Play soundfile before snswer   C F
• [Asterisk-Users] dtmf problem   Janina Sajka
• [Asterisk-Users] Play soundfile before snswer   C F
• [Asterisk-Users] How to check Digium TE410P firmware version?   C F
• [Asterisk-Users] spandsp & fax   Carlos Alperin
• [Asterisk-Users] UK, Disconnect supervision   Jonathan Attwood
• [Asterisk-Users] polycom sip slower than grandstream   Jerry Jones
• [Asterisk-Users] PRI: This number has been disconnected   Javier Ergas
• [Asterisk-Users] Asterisk Hosting   BILL GITONGA
• [Asterisk-Users] Asterisk Hosting   Michiel van Baak
• [Asterisk-Users] Blackberry SIM card   Chris Bagnall
• [Asterisk-Users] Blackberry SIM card   Alexander Lopez
• [Asterisk-Users] PRI: This number has been disconnected   Francesco Peeters (Asterisk)
• [Asterisk-Users] Automatic logoff of all agents at set time   Chuck Bunn
• [Asterisk-Users] Automatic logoff of all agents at set time   Michiel van Baak
• [Asterisk-Users] Automatic logoff of all agents at set time   Chuck Bunn
• [Asterisk-Users] Automatic logoff of all agents at set time   Michiel van Baak
• [Asterisk-Users] Automatic logoff of all agents at set time   Bud Bach
• [Asterisk-Users] Difference between CDR dispositions..   El Flynn
• [Asterisk-Users] [Announce] Pending Web-MeetMe update   Dan Austin
• [Asterisk-Users] Voicemail as other format?   Kerry Garrison
• [Asterisk-Users] Stay away from Grandstream!   isamar at isamarmaia.org
• [Asterisk-Users] Asterisk Christmas Help request   Mr Asterisk
• [Asterisk-Users] Blackberry SIM card   Jason p
• [Asterisk-Users] Stay away from Grandstream!   Erick Baum
• [Asterisk-Users] Stay away from Grandstream!   Elene Kinsky
• [Asterisk-Users] Asterisk seg fault (SVN-branch-1.2-r7641)   Robert La Ferla
• [Asterisk-Users] 4-port external sip fxo which doesnt suck?   asterisk at anime.net
• [Asterisk-Users] Stay away from Grandstream!   Steve Underwood
• [Asterisk-Users] 4-port external sip fxo which doesnt suck?   Chris Mason (Lists)
• [Asterisk-Users] Maximizing audio quality   Wolfgang Borgon
• [Asterisk-Users] Trial Edition of Druid Asterisk Web-interface   Vikram Rangnekar
• [Asterisk-Users] Asterisk on VPS   Dualcall.com
• [Asterisk-Users] Re: 4-port external sip fxo which doesnt suck?   Bob Knight
• [Asterisk-Users] Stay away from Grandstream!   calvis
• [Asterisk-Users] Stay away from Grandstream!   Brian Capouch
• [Asterisk-Users] Re: Cisco dtmf   Tomislav Parcina
• [Asterisk-Users] PHP Manager   Code Lover
• [Asterisk-Users] Maximizing audio quality   El Flynn
• [Asterisk-Users] Asterisk Hosting   Simon Woodhead
• [Asterisk-Users] Bad Checksum answering inbound call   Darren Younger
• [Asterisk-Users] billing system   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Re: Voicemail as other format?   Tomislav Parcina
• [Asterisk-Users] Re: Re: Grandstream Budge Tone 102   Tomislav Parcina
• [Asterisk-Users] Re: Does broadvoice modify caller ID name?   Tomislav Parcina
• [Asterisk-Users] Re: Transfer   Tomislav Parcina
• [Asterisk-Users] Re: 4-port external sip fxo which doesnt suck?   Rich Adamson
• [Asterisk-Users] Automatic logoff of all agents at set time   Michiel van Baak
• [Asterisk-Users]CLI execute extensions   asterisk183
• [Asterisk-Users] Stay away from Grandstream!   Avi Miller
• [Asterisk-Users] Wrong Password?????   Rafael Ledesma
• [Asterisk-Users] asterisk AVM C2 again   Dave Cotton
• [Asterisk-Users] Re: Why no sound from festival?   Tomislav Parcina
• [Asterisk-Users] asterisk AVM C2 again   stephane plichon
• [Asterisk-Users] Re: Asterisk Christmas Help request   Tomislav Parcina
• [Asterisk-Users] Re: channel monitoring whisper mode?   Tomislav Parcina
• [Asterisk-Users] agent logs   Dov Bigio
• [Asterisk-Users] spandsp & fax   Dov Bigio
• [Asterisk-Users] ISDN/CAPI outgoing calls - weirdness with ringing   Michael J. Tubby G8TIC
• [Asterisk-Users] UK, Disconnect supervision   Peter Hoppe
• [Asterisk-Users] unable to make calls out using ast@home   Peter Hoppe
• [Asterisk-Users] Stay away from Grandstream!   Vahan Yerkanian
• [Asterisk-Users] 4-port external sip fxo which doesnt suck?   Vahan Yerkanian
• [Asterisk-Users] UK, Disconnect supervision   Peter Hoppe
• [Asterisk-Users] Stay away from Grandstream!   Jacques Leisy
• [Asterisk-Users] Stay away from Grandstream!   Nir Simionovich
• [Asterisk-Users] UK, Disconnect supervision   Jonathan Attwood
• [Asterisk-Users] Sipura 2002 codec preferences   Chris Mason (Lists)
• [Asterisk-Users] Automatic logoff of all agents at set time   Chuck Bunn
• [Asterisk-Users] Re: Why no sound from festival?   Rich Adamson
• [Asterisk-Users] Dialling out with clone X100P board   Roger Hill
• [Asterisk-Users] Stay away from Grandstream!   Steve Underwood
• [Asterisk-Users] Automatic logoff of all agents at set time   Alexander Lopez
• [Asterisk-Users] Stay away from Grandstream!   Andrew Kohlsmith
• [Asterisk-Users] UK, Disconnect supervision   peter
• [Asterisk-Users] ipVolution   Goran Skular
• [Asterisk-Users] Strange IAX messages on the console   Dinesh Nair
• [Asterisk-Users] 4-port external sip fxo which doesnt suck?   Rich Adamson
• [Asterisk-Users] Automatic logoff of all agents at set time   Chuck Bunn
• [Asterisk-Users] Sipura 2002 codec preferences   Rich Adamson
• [Asterisk-Users] Automatic logoff of all agents at set time   Kevin P. Fleming
• [Asterisk-Users] oh323 configuration   Guillermo Salas M
• [Asterisk-Users] voip-info: Asterisk record calls   Tomislav Parcina
• [Asterisk-Users] oh323 configuration   Guillermo Salas M
• [Asterisk-Users] Automatic logoff of all agents at set time   Chuck Bunn
• [Asterisk-Users] Re: voip-info: Asterisk record calls   Tomislav Parcina
• [Asterisk-Users] PHP Manager   Moises Silva
• [Asterisk-Users] Sipura 2002 codec preferences   Carlos Alperin
• [Asterisk-Users] who is online   Kevin P. Fleming
• [Asterisk-Users] call transfer   Michael Sampson
• [Asterisk-Users] Wrong Password?????   Leah Newmark
• [Asterisk-Users] Stay away from Grandstream!   Nir Simionovich
• [Asterisk-Users] Re: who is online   Adrian Carter
• [Asterisk-Users] Re: who is online   bails
• [Asterisk-Users] Re: who is online   Douglas Garstang
• [Asterisk-Users] Blackberry SIM card   Robert Rawlinson
• [Asterisk-Users] Stay away from Grandstream!   Bjorn Asmul
• [Asterisk-Users] Cisco 7912G through NAT, problems with tones detection.   Hermann Wecke
• [Asterisk-Users] who is online   Pablo Allietti
• [Asterisk-Users] CallerID info needed   Doug Lytle
• [Asterisk-Users] call test   hgaillac-sip at yahoo.fr
• [Asterisk-Users] MWI problem   Il Neofita
• [Asterisk-Users] Re: who is online   Pablo Allietti
• [Asterisk-Users] What setup   John Crew
• [Asterisk-Users] BUG? AGI stuck in ast_waitfor_nandfds()   Moises Silva
• [Asterisk-Users] What setup   John Crew
• [Asterisk-Users] CALLERIDNUM   turby at seznam.cz
• [Asterisk-Users] What setup   John Crew
• [Asterisk-Users] Driver not configuring correctly on TE210P for CCS   Alex Barnes
• [Asterisk-Users] What setup   Cory Andrews
• [Asterisk-Users] CALLERIDNUM   C F
• [Asterisk-Users] PRI: This number has been disconnected   Javier Ergas
• [Asterisk-Users] Re: [Solved] who is online   Pablo Allietti
• Tr: Re: [Asterisk-Users] call test   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Most Stable Version of Asterisk   John Bittner
• [Asterisk-Users] CALLERIDNUM   turby at seznam.cz
• [Asterisk-Users] What setup   C F
• [Asterisk-Users] CALLERIDNUM   C F
• [Asterisk-Users] call transfer   Michael Sampson
• [Asterisk-Users] CallerID info needed   C F
• [Asterisk-Users] What setup   Ross C
• [Asterisk-Users] PRI outgoing caller ID stopped working   C F
• [Asterisk-Users] SIP to SIP calls   hgaillac-sip at yahoo.fr
• [Asterisk-Users] CallerID info needed   Doug Lytle
• [Asterisk-Users] Re: 26. RE: Stay away from Grandstream! (Bjorn Asmul)   Joe McConnaughey
• [Asterisk-Users] Re: Voicemail as other format?   Mark Quitoriano
• [Asterisk-Users] GSM-gateway setup   bbench at mail.bg
• [Asterisk-Users] subscription   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Sipura 2002 codec preferences   Kristian Kielhofner
• [Asterisk-Users] Re: who is online   Francesco Peeters (Asterisk)
• [Asterisk-Users] Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem   C F
• [Asterisk-Users] Iterfacing with a Mitel PBX   Tom Conklin
• [Asterisk-Users] how to alter cdr dst info?   Steven
• [Asterisk-Users] Asterisk in Spanish   Erick Perez
• [Asterisk-Users] Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem   Rich Adamson
• [Asterisk-Users] Problems with multiple outbound calls going to PSTN - Wildcard TE405P   S. Dale
• [Asterisk-Users] Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem   C F
• [Asterisk-Users] Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem   C F
• [Asterisk-Users] extension not ringing when dialed from DID   Leonard Burton
• [Asterisk-Users] spool calls always failing with congestion (AST_CONTROL_CONGESTION)   Technical Support
• [Asterisk-Users] CALLERIDNUM and SRC in CDR   turby at seznam.cz
• [Asterisk-Users] CALLERIDNUM and SRC in CDR   C F
• [Asterisk-Users] CALLERIDNUM and SRC in CDR   turby at seznam.cz
• [Asterisk-Users] Asterisk as a Gateway   Nitesh Divecha
• [Asterisk-Users] Static (Distortion?) and noise on FXO (TDM04b)   Pete Doyle
• [Asterisk-Users] sip debug > file.txt   Leonard Burton
• [Asterisk-Users] call test   aturntablist
• [Asterisk-Users] Asterisk as a Gateway   BJ Weschke
• [Asterisk-Users] Re: CALLERIDNUM and SRC in CDR   Steven
• [Asterisk-Users] Most Stable Version of Asterisk   Jason Becker
• [Asterisk-Users] Soundstation 4000   Chris Gamble
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   Phil Pritchard
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   Kerry Garrison
• [Asterisk-Users] sip debug > file.txt   Tzafrir Cohen
• [Asterisk-Users] 4-port external sip fxo which doesnt suck?   Michael Graves
• [Asterisk-Users] Stay away from Grandstream!   Chris Bagnall
• [Asterisk-Users]CLI execute extensions   Tzafrir Cohen
• [Asterisk-Users] Polycom check-sync   Jerry Jones
• [Asterisk-Users] Polycom check-sync   BJ Weschke
• [Asterisk-Users] Polycom check-sync   Kevin P. Fleming
• [Asterisk-Users] Polycom check-sync   Douglas Garstang
• [Asterisk-Users] Most Stable Version of Asterisk   Matt Florell
• [Asterisk-Users] Asterisk as a Gateway   Nitesh Divecha
• [Asterisk-Users] Regular crashes   Andrew Gough
• [Asterisk-Users] Driver not configuring correctly on TE210P for CCS   Paul Liew
• [Asterisk-Users] PRI: This number has been disconnected   Joe Pukepail
• [Asterisk-Users] Soundstation 4000   BJ Weschke
• [Asterisk-Users] Polycom check-sync   Jerry Jones
• [Asterisk-Users] Conditional CODEC translation   Leandro Rzezak
• [Asterisk-Users] Asterisk as a Gateway   James Sizemore
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   Kristian Kielhofner
• [Asterisk-Users] Soundstation 4000   Carlos Chavez
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   Kerry Garrison
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   William Boehlke
• [Asterisk-Users] Asterisk as a Gateway   Nitesh Divecha
• [Asterisk-Users] Asterisk as a Gateway   BJ Weschke
• [Asterisk-Users] Iterfacing with a Mitel PBX   Mark Phillips
• [Asterisk-Users] TDM2400 wierdness   Kerry Garrison
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   Kerry Garrison
• [Asterisk-Users] CALLERIDNUM   Rehan Ahmed
• [Asterisk-Users] oh323 configuration   Rehan Ahmed
• [Asterisk-Users] spandsp & fax   Rehan Ahmed
• [Asterisk-Users] Asterisk as a Gateway   Nitesh Divecha
• [Asterisk-Users] spandsp & fax   Carlos Alperin
• [Asterisk-Users] oh323 configuration   Guillermo Salas M
• [Asterisk-Users] ipVolution   Greg Boehnlein
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   William Boehlke
• [Asterisk-Users] extension not ringing when dialed from DID   Eric "ManxPower" Wieling
• [Asterisk-Users] Asterisk as a Gateway   Ray Yang
• [Asterisk-Users] Re: Re: How to record a call   Dan Littlejohn
• [Asterisk-Users] Conditional CODEC translation   Leandro Rzezak
• [Asterisk-Users] select codec based on extension   Leandro Rzezak
• [Asterisk-Users] Asterisk as a Gateway   Nitesh Divecha
• [Asterisk-Users] Go directly to new messages from VoiceMailMain?   Robert La Ferla
• [Asterisk-Users] Grandstream Configuration Utility available   Cory Andrews
• [Asterisk-Users] Stay away from Grandstream!   isamar at isamarmaia.org
• [Asterisk-Users] Linksys SPA-9000   Kerry Garrison
• [Asterisk-Users] TDM2400 wierdness   Kerry Garrison
• [Asterisk-Users] Linksys SPA-9000   Cory Andrews
• [Asterisk-Users] Re: What setup   John Crew
• [Asterisk-Users] Bad Checksum answering inbound call   pdhales at optusnet.com.au
• [Asterisk-Users] Re: Re: Voicemail as other format?   Tomislav Parcina
• [Asterisk-Users] I thought they weren't charging - FW: [DIDx.net] Happy holidays wishes from DIDX.net.   Don Fanning
• [Asterisk-Users] select codec based on extension   Simone Cittadini
• [Asterisk-Users] Bad Checksum answering inbound call   Darren Younger
• [Asterisk-Users] TDM2400 wierdness   Roger Hill
• [Asterisk-Users] Bad Checksum answering inbound call   pdhales at optusnet.com.au
• [Asterisk-Users] select codec based on extension   pdhales at optusnet.com.au
• [Asterisk-Users] Bad Checksum answering inbound call   pdhales at optusnet.com.au
• [Asterisk-Users] Re: Problems with multiple outbound calls going to PSTN - Wildcard TE405P   Tomislav Parcina
• [Asterisk-Users] select codec based on extension   Francesco Peeters (Asterisk)
• [Asterisk-Users] call test   hgaillac-sip at yahoo.fr
• [Asterisk-Users] call test   Peter Bowyer
• [Asterisk-Users] Re: Problems with multiple outbound calls going toPSTN - Wildcard TE405P   pdhales at optusnet.com.au
• [Asterisk-Users] What is the best Dell Machine for Asterisk?   Jens Vagelpohl
• [Asterisk-Users] s or _X. , is there any change since Asterisk 1.2   Zeeshan
• [Asterisk-Users] HELP! Asterisk 1.2.1 stops immediately - voicemail problem?   Kib Eki
• [Asterisk-Users] PRI: This number has been disconnected   Adam Goryachev
• [Asterisk-Users] Congestion problem   Tomislav Parcina
• [Asterisk-Users] SIP to SIP calls   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Conditional CODEC translation   Rehan AllahWala
• [Asterisk-Users] sip debug > file.txt   Olle E Johansson
• [Asterisk-Users] Driver not configuring correctly on TE210P forCCS   Alex Barnes
• [Asterisk-Users] SIP to SIP calls   Rehan AllahWala
• [Asterisk-Users] Re: Go directly to new messages from VoiceMailMain?   Tomislav Parcina
• [Asterisk-Users] Re: Grandstream Configuration Utility available   Tomislav Parcina
• [Asterisk-Users] I thought they weren't charging - FW: [DIDx.net] Happy holidays wishes from DIDX.net.   Linda Goldsmith
• [Asterisk-Users] Re: What is the best Dell Machine for Asterisk?   Steven
• [Asterisk-Users] Re: Re: Problems with multiple outbound calls going toPSTN - Wildcard TE405P   Tomislav Parcina
• [Asterisk-Users] spandsp & fax   Rehan AllahWala
• [Asterisk-Users] HELP! Asterisk 1.2.1 stops immediately - voicemail problem? - SOLVED   Kib Eki
• [Asterisk-Users] TDM2400 wierdness   Kerry Garrison
• [Asterisk-Users] Conditional CODEC translation   Bogdan Moldovan
• [Asterisk-Users] Aastra firmware 1.3.x   Michael George
• [Asterisk-Users] SNOM 360 locked up SOLVED   Michael George
• [Asterisk-Users] SNOM 360 locked up   Sven Fischer (support)
• [Asterisk-Users] SPA-3000 + call waiting   Ugo Bellavance
• [Asterisk-Users] smsq   chris songer
• [Asterisk-Users] SNOM 360 locked up SOLVED   Christian Stredicke
• [Asterisk-Users] Easiest way to use HFC-S?   Pisac
• [Asterisk-Users] Asterisk as a Gateway   James Sizemore
• [Asterisk-Users] Asterisk as a Gateway   James Sizemore
• [Asterisk-Users] SPA-3000 + call waiting   Kerry Garrison
• [Asterisk-Users] smsq   Jens Vagelpohl
• [Asterisk-Users] TDM2400 wierdness   Tom Vile
• [Asterisk-Users] Re: Go directly to new messages from VoiceMailMain?   Robert La Ferla
• [Asterisk-Users] Re: Go directly to new messages fromVoiceMailMain?   Alexander Lopez
• [Asterisk-Users] Asterisk Server Hangs   Dushyanth Harinath
• [Asterisk-Users] zaptel TDM21B 4-5 second pause   Eck
• [Asterisk-Users] What does "Page" application do?   Robert La Ferla
• [Asterisk-Users] Linksys SPA-9000   Robert Augustyn
• [Asterisk-Users] Linksys SPA-9000   Kerry Garrison
• [Asterisk-Users] CoreDump   Markus Monka
• [Asterisk-Users] billing system   janvb at caselaboratories.com
• [Asterisk-Users] Asterisk 1.2 + DMZ + NAT clients   Blake Krone
• [Asterisk-Users] What does "Page" application do?   janvb at caselaboratories.com
• [Asterisk-Users] Asterisk SIP PORTS   Kanishka Somaratne
• [Asterisk-Users] What does "Page" application do?   Andrew Latham
• [Asterisk-Users] Asterisk 1.2 + DMZ + NAT clients   Kerry Garrison
• [Asterisk-Users] PRI Hangup cause   Kevin Bockman
• [Asterisk-Users] Asterisk 1.2 + DMZ + NAT clients   Blake Krone
• [Asterisk-Users] CALLERIDNUM   C F
• [Asterisk-Users] PRI: This number has been disconnected   Joe Pukepail
• [Asterisk-Users] What does "Page" application do?   Robert La Ferla
• [Asterisk-Users] Re: Go directly to new messages fromVoiceMailMain?   BJ Weschke
• [Asterisk-Users] Problem getting D channel up on Sangoma A102   courchea at net-forces.com
• [Asterisk-Users] Gnet VP168S   Tim Johnson
• [Asterisk-Users] TDM2400 wierdness   Kerry Garrison
• [Asterisk-Users] PRI: This number has been disconnected   Javier Ergas
• [Asterisk-Users] Realtime Multiple Asterisk boxes and rtcachefriends MWI   Asterisk
• [Asterisk-Users] fernando_ldb@yahoo.com.ar   Fernando Lopez de Briÿfffffffffff1as
• [Asterisk-Users] What does "Page" application do?   Andrew Latham
• [Asterisk-Users] PRI: This number has been disconnected   Javier Ergas
• [Asterisk-Users] Realtime Multiple Asterisk boxes andrtcachefriends MWI   Douglas Garstang
• [Asterisk-Users] What does "Page" application do?   Jacques Leisy
• [Asterisk-Users] spandsp & fax   Carlos Alperin
• [Asterisk-Users] spandsp & fax   Pierre Burton
• [Asterisk-Users] Linksys SPA-942   Hunt, Bill
• [Asterisk-Users] spandsp & fax   Carlos Alperin
• [Asterisk-Users] Asterisk 1.2 + DMZ + NAT clients   Hadley Rich
• [Asterisk-Users] Linksys SPA-942   Cory Andrews
• [Asterisk-Users] spandsp & fax   Dov Bigio
• [Asterisk-Users] Linksys SPA-942   Kerry Garrison
• [Asterisk-Users] Regular modems?   Matt Murphy
• [Asterisk-Users] spandsp & fax   Carlos Alperin
• [Asterisk-Users] voicemail storage over odbc and postgres   Elazar Rosenthal
• [Asterisk-Users] Getting Yoda unit to register all four ports   Chris Mason (Lists)
• [Asterisk-Users] Asterisk SIP PORTS   Rehan AllahWala
• [Asterisk-Users] zaptel TDM21B 4-5 second pause   Giovanni Miano
• [Asterisk-Users] sip debug > file.txt   Tzafrir Cohen
• [Asterisk-Users] Asterisk Server Hangs   Giovanni Miano
• [Asterisk-Users] Easiest way to use HFC-S?   Giovanni Miano
• [Asterisk-Users] transfers using # in asterisk   Franklin Webb
• [Asterisk-Users] Regular modems?   John Novack
• [Asterisk-Users] Asterisk as a Gateway   Nitesh Divecha
• [Asterisk-Users] Need Asterisk person in SoCal   Kerry Garrison
• [Asterisk-Users] Asterisk as a Gateway   James Sizemore
• [Asterisk-Users] Regular modems?   Matt Murphy
• [Asterisk-Users] Allison on Free 411   Joe Pukepail
• [Asterisk-Users] PRI: This number has been disconnected   Joe Pukepail
• [Asterisk-Users] Asterisk as a Gateway   pdhales at optusnet.com.au
• [Asterisk-Users] transfers using # in asterisk   pdhales at optusnet.com.au
• [Asterisk-Users] What does "Page" application do?   janvb at caselaboratories.com
• [Asterisk-Users] Regular modems?   Kerry Garrison
• [Asterisk-Users] zaptel TDM21B 4-5 second pause   Eck
• [Asterisk-Users] voicemail storage over odbc and postgres   Michiel van Baak
• [Asterisk-Users] Allison on Free 411   Mark Phillips
• [Asterisk-Users] transfers using # in asterisk   trixter aka Bret McDanel
• [Asterisk-Users] CALLERIDNUM   Rehan AllahWala
• [Asterisk-Users] Allison on Free 411   Dean Collins
• [Asterisk-Users] Allison on Free 411   Kerry Garrison
• [Asterisk-Users] Easiest way to use HFC-S?   Pisac
• [Asterisk-Users] Re: Stay away from Grandstream!   Wolfgang S. Rupprecht
• [Asterisk-Users] Re: Stay away from Grandstream!   Kerry Garrison
• [Asterisk-Users] Re: Stay away from Grandstream!   Andrew Kohlsmith
• [Asterisk-Users] Semi-OT: porting numbers away   Ross C
• [Asterisk-Users] SetAccount missing?   William M. Sandiford
• [Asterisk-Users] Sending Polarity/DTMF Caller ID in chan_zap (Sweden etc...)   Josef Seger
• [Asterisk-Users] Semi-OT: porting numbers away   Kerry Garrison
• [Asterisk-Users] Allison on Free 411   Mark Phillips
• [Asterisk-Users] Semi-OT: porting numbers away   Ross C
• [Asterisk-Users] PRI: This number has been disconnected   Adam Goryachev
• [Asterisk-Users] silent dial/ring?   Robert La Ferla
• [Asterisk-Users] Re: Go directly to new messagesfromVoiceMailMain?   Alexander Lopez
• [Asterisk-Users] Re: Go directly to new messagesfromVoiceMailMain?   Robert La Ferla
• [Asterisk-Users] What does "Page" application do?   Robert La Ferla
• [Asterisk-Users] Semi-OT: porting numbers away   Leonard Burton
• [Asterisk-Users] SetAccount missing?   Robert La Ferla
• [Asterisk-Users] What does "Page" application do?   Alexander Lopez
• [Asterisk-Users] CALLERIDNUM   Alexander Lopez
• [Asterisk-Users] Semi-OT: porting numbers away   Ross C
• [Asterisk-Users] RE:probelem in working of X100P   Tejas Shah
• [Asterisk-Users] Easiest way to use HFC-S?   Peer Oliver Schmidt
• [Asterisk-Users] Re: Stay away from Grandstream!   BennyBad
• [Asterisk-Users] 4-port external sip fxo which doesnt suck?   George Pajari
• [Asterisk-Users] Re: Congestion problem   Tomislav Parcina
• [Asterisk-Users] Re: Congestion problem   Brian Capouch
• [Asterisk-Users] Re: CALLERIDNUM (Rehan AllahWala)   Freddi Hansen
• [Asterisk-Users] sip debug > file.txt   Olle E Johansson
• [Asterisk-Users] CALLERIDNUM   Rehan AllahWala
• [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers   Simone Cittadini
• [Asterisk-Users] Outbound call using ISDN extension disconnected after *exactly* 30 seconds   Francesco Peeters (Asterisk)
• [Asterisk-Users] SetAccount missing?   Michiel van Baak
• [Asterisk-Users] Asterisk connect to voicemaster configuration 1.7   Angelito Manansala
• [Asterisk-Users] Howto config tdm2400   Manuel Casal
• [Asterisk-Users] Can we dial agents from extensions.conf   vivek at staff.ownmail.com
• [Asterisk-Users] Can we dial agents from extensions.conf   Alexander Lopez
• [Asterisk-Users] Howto config tdm2400   BJ Weschke
• [Asterisk-Users] Howto config tdm2400   f6hqz-m at hamwlan.net
• [Asterisk-Users] Semi-OT: porting numbers away   Rich Adamson
• [Asterisk-Users] Howto config tdm2400   Manuel Casal
• [Asterisk-Users] Queue features   Dov Bigio
• [Asterisk-Users] Semi-OT: porting numbers away   Austin Denyer
• [Asterisk-Users] Problem getting D channel up on Sangoma A102   Rich Adamson
• [Asterisk-Users] Queue features   Michiel van Baak
• [Asterisk-Users] SetAccount missing?   Andrew Latham
• [Asterisk-Users] Queue features   Giovanni Miano
• [Asterisk-Users] Howto config tdm2400   BJ Weschke
• [Asterisk-Users] Problem on ZAP channel   rbrahmbhatt at adiance.com
• [Asterisk-Users] Can we dial agents from extensions.conf   vivek at staff.ownmail.com
• [Asterisk-Users] Problem getting D channel up on Sangoma A102   David Yat Sin
• [Asterisk-Users] Regular Crashes   Andrew Gough
• [Asterisk-Users] Problem on ZAP channel   Michael Sampson
• [Asterisk-Users] Queue features   Dov Bigio
• [Asterisk-Users] wctdm module goes missing after a reboot - Gentoo?   Ryan Booz
• [Asterisk-Users] Semi-OT: porting numbers away   Julio Arruda
• [Asterisk-Users] Regular Crashes   Zafer Khodr
• [Asterisk-Users] Semi-OT: porting numbers away   Rich Adamson
• [Asterisk-Users] wctdm module goes missing after a reboot - Gentoo?   Moises Silva
• [Asterisk-Users] PRI: This number has been disconnected   Javier Ergas
• [Asterisk-Users] Asterisk connect to voicemaster configuration 1.7   Moises Silva
• [Asterisk-Users] No RTP Warning   William M. Sandiford
• [Asterisk-Users] MYSQL Fetch Warning   William M. Sandiford
• [Asterisk-Users] Asterisk connect to voicemaster configuration 1.7   Bill Gibbs
• [Asterisk-Users] SNOM 360 locked up SOLVED   Janina Sajka
• [Asterisk-Users] Semi-OT: porting numbers away   Kerry Garrison
• [Asterisk-Users] TBCT For PRI support   Chris Matthews
• [Asterisk-Users] Has anyone used the applicationmap in features.conf?   John Voss
• [Asterisk-Users] Problem on ZAP channel   Simone Cittadini
• [Asterisk-Users] Semi-OT: porting numbers away   Rich Adamson
• [Asterisk-Users] Notifications when host fails qualify   Jonathan k. Creasy
• [Asterisk-Users] using a Gigaset SX440isdn on a Diva 4BRI ?   Louis-David Mitterrand
• [Asterisk-Users] RPID Issue   Ray Van Dolson
• [Asterisk-Users] TDM400 FXO outbound issue   Jason D. Wolfe
• [Asterisk-Users] FOP Maximum extensions?   Dan Elder
• [Asterisk-Users] FOP Maximum extensions?   Kerry Garrison
• [Asterisk-Users] IAX problem - Bug or Compatibility issue?   Aryanto Rachmad
• [Asterisk-Users] FOP Maximum extensions?   Douglas Garstang
• [Asterisk-Users] Playback after Page()   Douglas Garstang
• [Asterisk-Users] PRI: This number has been disconnected   Andres
• [Asterisk-Users] Semi-OT: porting numbers away   Ross C
• [Asterisk-Users] call sip:info@nxs.yi.org   hgaillac-sip at yahoo.fr
• [Asterisk-Users] Vonage Sip Peering   Ben Higley
• [Asterisk-Users] Manually Opening and Closing a Queue   Bud Bach
• [Asterisk-Users] Semi-OT: porting numbers away   trixter aka Bret McDanel
• [Asterisk-Users] ENUM trees   Joao Pereira
• [Asterisk-Users] Playback after Page()   C F
• [Asterisk-Users] ENUM trees   trixter aka Bret McDanel
• [Asterisk-Users] Semi-OT: porting numbers away   Ross C
• [Asterisk-Users] Semi-OT: porting numbers away   Andrew Kohlsmith
• [Asterisk-Users] Problem on ZAP channel   rbrahmbhatt at adiance.com
• [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 176   Bill Michaelson
• [Asterisk-Users] Semi-OT: porting numbers away   Matt Riddell
• [Asterisk-Users] Fax Support   rbrahmbhatt at adiance.com
• [Asterisk-Users] Semi-OT: porting numbers away   Ross C
• [Asterisk-Users] Semi-OT: porting numbers away   Bogdan Moldovan
• [Asterisk-Users] Which Asterisk GUI?   Ken D'Ambrosio
• [Asterisk-Users] Semi-OT: porting numbers away   trixter aka Bret McDanel
• [Asterisk-Users] Semi-OT: porting numbers away   Brian Capouch
• [Asterisk-Users] Semi-OT: porting numbers away   trixter aka Bret McDanel
• [Asterisk-Users] Semi-OT: porting numbers away   trixter aka Bret McDanel
• [Asterisk-Users] Semi-OT: porting numbers away   Jerry Jones
• [Asterisk-Users] Semi-OT: porting numbers away   Jerry Jones
• [Asterisk-Users] Which Asterisk GUI?   Ariel Batista
• [Asterisk-Users] Semi-OT: porting numbers away   John Novack
• [Asterisk-Users] Semi-OT: porting numbers away   Bogdan Moldovan
• [Asterisk-Users] Semi-OT: porting numbers away   Ron Wellsted
• [Asterisk-Users] Problem on ZAP channel   Steve Totaro
• [Asterisk-Users] Passing authentication to an analog adapter   Philip Edelbrock
• [Asterisk-Users] Which Asterisk GUI?   George Pajari
• [Asterisk-Users] Semi-OT: porting numbers away   Steve Kennedy
• [Asterisk-Users] Cheap FXS/USB terminal SE-B2K, can it work with asterisk?   Dan Elder
• [Asterisk-Users] Motherboard choice for large opteron based asterisk server?   Mike Fedyk
• [Asterisk-Users] Aterisk 1.2.1 zaptel module not found   jonny hashem
• [Asterisk-Users] NOOB: Need Help Learning How to Debug PRI (U.S.)   Michael Collins
• [Asterisk-Users] using a Gigaset SX440isdn on a Diva 4BRI ?   Armin Schindler
• [Asterisk-Users] Recording Calls for Specific ACD Agents   Douglas Garstang
• [Asterisk-Users] Can we dial agents from extensions.conf   Alexander Lopez
• [Asterisk-Users] Aterisk 1.2.1 zaptel module not found   Moises Silva
• [Asterisk-Users] Re: using a Gigaset SX440isdn on a Diva 4BRI ?   Armin Schindler
• [Asterisk-Users] Semi-OT: porting numbers away   Rich Adamson
• [Asterisk-Users] Playback after Page()   Alexander Lopez
• [Asterisk-Users] voicemail .wav filename   Michiel van Baak
• [Asterisk-Users] Outputting human readable info on a VoIP call's quality?   S McGowan
• [Asterisk-Users] Aterisk 1.2.1 zaptel module not found   f6hqz-m at hamwlan.net
• [Asterisk-Users] Semi-OT: porting numbers away   trixter aka Bret McDanel
• [Asterisk-Users] Outputting human readable info on a VoIP call's quality?   BJ Weschke
• [Asterisk-Users] Recording Calls for Specific ACD Agents   BJ Weschke
• [Asterisk-Users] Recording Calls for Specific ACD Agents   Douglas Garstang
• [Asterisk-Users] Aterisk 1.2.1 zaptel module not found   chawki hammoud
• [Asterisk-Users] Aterisk 1.2.1 zaptel module not found   f6hqz-m at hamwlan.net
• [Asterisk-Users] GXP-2000 any good with * ?   asterisk at anime.net
• [Asterisk-Users] voicemail .wav filename   Alexander Lopez
• [Asterisk-Users] Call Groups   Douglas Garstang
• [Asterisk-Users] Problem on ZAP channel   Javier Ergas
• [Asterisk-Users] Aterisk 1.2.1 zaptel module not found   chawki hammoud
• [Asterisk-Users] GXP-2000 any good with * ?   trixter aka Bret McDanel
• [Asterisk-Users] GXP-2000 any good with * ?   Michiel van Baak
• [Asterisk-Users] Re: SPA-3000 + call waiting   Ugo Bellavance
• [Asterisk-Users] GXP-2000 any good with * ?   Tom Vile
• [Asterisk-Users] sip through nat problem   Alyed Tzompa
• [Asterisk-Users] GXP-2000 any good with * ?   Michiel van Baak
• [Asterisk-Users] GXP-2000 any good with * ?   Ross C
• [Asterisk-Users] GXP-2000 any good with * ?   Tom Vile
• [Asterisk-Users] GXP-2000 any good with * ?   Ken D'Ambrosio
• [Asterisk-Users] IAX problem - Bug or Compatibility issue?   Aryanto Rachmad
• [Asterisk-Users] Best way to terminate calls   Tom Vile
• [Asterisk-Users] GXP-2000 any good with * ?   Bjorn Asmul
• [Asterisk-Users] GXP-2000 any good with * ?   Michiel van Baak
• [Asterisk-Users] Cheap FXS/USB terminal SE-B2K, can it work with asterisk?   Mark Phillips
• [Asterisk-Users] GXP-2000 any good with * ?   Ross C
• [Asterisk-Users] GXP-2000 any good with * ?   Tom Vile
• [Asterisk-Users] Semi-OT: porting numbers away   gw at adcomcorp.com
• [Asterisk-Users] Dicate not completing the DialPlan?   Bob Weber
• [Asterisk-Users] name that vendor...   asterisk at anime.net
• [Asterisk-Users] GXP-2000 any good with * ?   Michiel van Baak
• [Asterisk-Users] GXP-2000 any good with * ?   Tom Vile
• [Asterisk-Users] wctdm module goes missing after a reboot - Gentoo?   Robert La Ferla
• [Asterisk-Users] name that vendor...   trixter aka Bret McDanel
• [Asterisk-Users] name that vendor...   Mark Phillips
• [Asterisk-Users] Re: Go directly to new messagesfromVoiceMailMain?   Robert La Ferla
• [Asterisk-Users] name that vendor...   Chris Mason
• [Asterisk-Users] name that vendor...   Robert La Ferla
• [Asterisk-Users] name that vendor...   Cory Andrews
• [Asterisk-Users] Easiest way to use HFC-S?   Pisac
• [Asterisk-Users] Re: Go directly to new messagesfromVoiceMailMain?   Alexander Lopez
• [Asterisk-Users] Outputting human readable info on a VoIP call'squality?   S McGowan
• [Asterisk-Users] RE:problem with X100P card   Tejas Shah
• [Asterisk-Users] Sip man in the middle   Mike Bernson
• [Asterisk-Users] Recording Calls for Specific ACD Agents   BJ Weschke
• [Asterisk-Users] Asterisk connect to voicemaster configuration 1.7   Angelito Manansala
• [Asterisk-Users] name that vendor...   asterisk at anime.net
• [Asterisk-Users] answer supervision and POTS   Jason D. Wolfe
• [Asterisk-Users] Can we dial agents from extensions.conf   vivek at staff.ownmail.com
• [Asterisk-Users] Short code to replace the confirmation key   Stephen Arulraj
• [Asterisk-Users] AGI Variable   Code Lover
• [Asterisk-Users] Multiple Realm Definitions?   S McGowan
• [Asterisk-Users] Aterisk 1.2.1 zaptel module not found   f6hqz-m at hamwlan.net
• [Asterisk-Users] name that vendor...   f6hqz-m at hamwlan.net
• [Asterisk-Users] RE:problem with X100P card   f6hqz-m at hamwlan.net
• [Asterisk-Users] Notifications when host fails qualify   Olle E Johansson
• [Asterisk-Users] RPID Issue   Olle E Johansson
• [Asterisk-Users] Multiple Realm Definitions?   Olle E Johansson
• [Asterisk-Users] Short code to replace the confirmation key   Stephen Arulraj
• [Asterisk-Users] name that vendor...   Jeffery Chen
• [Asterisk-Users] GXP-2000 any good with * ?   Kristof Hardy
• [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP   Peter Bowyer
• [Asterisk-Users] Easiest way to use HFC-S?   Peer Oliver Schmidt
• [Asterisk-Users] name that vendor...   f6hqz-m at hamwlan.net
• [Asterisk-Users] CAPI and *   Sascha Andres
• [Asterisk-Users] Sip man in the middle   Stewart Nelson
• [Asterisk-Users] CAPI and *   Armin Schindler
• [Asterisk-Users] [Announcement] chan_capi-cm 0.6.2 released   Armin Schindler
• [Asterisk-Users] How to check Queue Statistics   Zeeshan
• [Asterisk-Users] name that vendor...   Mark Phillips
• [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP   Kristof Hardy
• [Asterisk-Users] name that vendor...   Vahan Yerkanian
• [Asterisk-Users] name that vendor...   Matteo Brancaleoni
• [Asterisk-Users] name that vendor...   f6hqz-m at hamwlan.net
• [Asterisk-Users] CAPI and *   Sascha Andres
• [Asterisk-Users] CAPI and *   Armin Schindler
• [Asterisk-Users] How to set features.conf to change the hangup key.   Obelix
• [Asterisk-Users] CAPI and *   Sascha Andres
• [Asterisk-Users] How to set features.conf to change the hangup key.   Bogdan Moldovan
• [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP   Ross C
• [Asterisk-Users] CAPI and *   Armin Schindler
• [Asterisk-Users] How to set features.conf to change the hangup key.   Obelix
• [Asterisk-Users] Outputting human readable info on a VoIP call'squality?   Moises Silva
• [Asterisk-Users] How to check Queue Statistics   BJ Weschke
• [Asterisk-Users] How to set features.conf to change thehangup key.   Bogdan Moldovan
• [Asterisk-Users] How to set features.conf to change thehangup key.   Obelix
• [Asterisk-Users] How to set features.conf to change thehangup key.   Bogdan Moldovan
• [Asterisk-Users] Having major issues with TDM2400   Kerry Garrison
• [Asterisk-Users] name that vendor...   Chris Mason (Lists)
• [Asterisk-Users] voicemail/privacy system   Roy Kidder
• [Asterisk-Users] misconfigured autoresponder   trixter aka Bret McDanel
• [Asterisk-Users] voicemail/privacy system   Moises Silva
• [Asterisk-Users] wctdm module goes missing after a reboot - Gentoo?   Moises Silva
• [Asterisk-Users] SIP Subscription Storage Location   BJ Weschke
• [Asterisk-Users] How to check Queue Statistics   Zeeshan
• [Asterisk-Users] How to check Queue Statistics   BJ Weschke
• [Asterisk-Users] voicemail/privacy system   Eck
• [Asterisk-Users] Semi-OT: porting numbers away   Janina Sajka
• [Asterisk-Users] Semi-OT: porting numbers away   trixter aka Bret McDanel
• [Asterisk-Users] OT Maybe: Anyone have any knowledge of v5.1/v5.2 in connection with Asterisk?   Nir Simionovich
• [Asterisk-Users] tdm dialout delay   Diyanat Ali
• [Asterisk-Users] New Manager Client Program   Bill Michaelson
• [Asterisk-Users] Re: Fax Support   LJ
• [Asterisk-Users] Re: Fax Support   trixter aka Bret McDanel
• [Asterisk-Users] Need HT488 FXO example for both inbound and outbound.   James Ronald
• [Asterisk-Users] SPA-3K FXO: Incoming and outgoing calls in different contexts?   Brian Capouch
• [Asterisk-Users] Re: Semi-OT: porting numbers away   Wolfgang S. Rupprecht
• [Asterisk-Users] Voicemail through outlook   S.Ammad Jami