6,048 messages

com.digium.lists.asterisk-users [All Lists]

2005 January [All Months]

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[Asterisk-Users] call forwarding - mohammad
[Asterisk-Users] Eyebeam - asterisk - Messenger - Ferguson, Michael
[Asterisk-Users] Asterisk@home and Zap Channels - Chuck Keeter
[Asterisk-Users] RE: Q: Can I over-ride the value of ${CALLERIDNAME} ? - Brian Christie
[Asterisk-Users] Strange Crash - Steven Critchfield
[Asterisk-Users] Strange Crash - Paradise Dove
[Asterisk-Users] Asterisk on MS Virtual Server - Paul Tyreman
[Asterisk-Users] widcard x100P doubt - Lyle Giese
[Asterisk-Users] where to buy x100p - dean collins
[Asterisk-Users] Anyone having problems with LiveVoIP? - Tim Lewis
[Asterisk-Users] agent logoff - Joe Dennick
[Asterisk-Users] Record inbound and outbound calls to and from one number. - Mark Phillips
[Asterisk-Users] Silly question: Why multiple lines on SIP phones? - Michael Graves
[Asterisk-Users] Re: Sipura SPA-841 auto-answer support [patch] - Geoff Speicher
[Asterisk-Users] Sipura SPA-841 auto-answer support [patch] - Olle E. Johansson
[Asterisk-Users] Caller ID spoofing - Roger Schreiter
[Asterisk-Users] Caller ID spoofing - Calin Serbanescu
[Asterisk-Users] asterisk tries to dial out on lines already in use. - Paradise Dove
[Asterisk-Users] Polycom changing policy - allowing firmwaredownloads? - Mark Eissler
[Asterisk-Users] conference room capacity question - M.N.A.Smadi
[Asterisk-Users] Trying to make but it fails - hel...@per-s.dk
[Asterisk-Users] Monitor calls timeout - jurgen
[Asterisk-Users] asterisk tries to dial out on lines already in use. - Jon Gabrielson
[Asterisk-Users] New Firefly version - Adam Hart
[Asterisk-Users] New Firefly version - Duane
[Asterisk-Users] Asterisk friendly VoIP providers - Manjit Riat
[Asterisk-Users] Processing incoming calls with multiple contextst over PRI - Jason Brown
[Asterisk-Users] Processing incoming calls with multiple contextst over PRI - el Flynn
[Asterisk-Users] Processing incoming calls with multiple contextst over PRI - Kevin P. Fleming
[Asterisk-Users] Processing incoming calls with multiple contextstover PRI - Lyle Giese
[Asterisk-Users] Hitting IOCTL?? - Robert Webb
[Asterisk-Users] Asterisk@home and Zap Channels - Chuck Keeter
[Asterisk-Users] CAC Access Bank - Matt Riddell
[Asterisk-Users] how to stop ringing after congestion. - Jon Gabrielson
[Asterisk-Users] Broadvoice problems - Manjit Riat
[Asterisk-Users] Japan - Jason Frisch
[Asterisk-Users] Re: Record inbound and outbound calls to and from one number. - Tom Shoval
[Asterisk-Users] Japan - Leo Ann Boon
[Asterisk-Users] Re: Record inbound and outbound calls to and from one number. - Tim Mattison
[Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 crashes with Ouch ... error while writing audio data: : Broken pipe - Klaus-Peter Junghanns
[Asterisk-Users] Call Screen Macro Not Exiting when call rejected - RockWater !
[Asterisk-Users] Zap channels in AU hanging up on STD pips - Simon Brown
[Asterisk-Users] Monitor calls timeout - Trevor Peirce
[Asterisk-Users] Instant Messaging - Paolo Elefante
[Asterisk-Users] Zap channels in AU hanging up on STD pips - Gary
[Asterisk-Users] Group Extension - el Flynn
[Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 crashes with Ouch ... error while writing audio data: : Broken pipe - Remco Barende
AW: [Asterisk-Users] HDLC for Dummies? - Sebastian Buntin
[Asterisk-Users] SIP x NAT - César Davi Ávila do Nascimento
[Asterisk-Users] NAT and SIP - Rich Adamson
[Asterisk-Users] reason 24 (Call ended with Q.931 cause) - Michael Manousos
[Asterisk-Users] Caller ID in AU - Peter Illmayer
[Asterisk-Users] Fwd and Tollfree - Rene Kluwen
[Asterisk-Users] Trunked IAX or not - Mark Eissler
[Asterisk-Users] Asterisk with Grandstream ringback - Craig Guy
[Asterisk-Users] Announcement to caller when called party haspicked up - without initial Answer()? - Alex Barnes
[Asterisk-Users] Announcement to caller when called party has picked up - without initial Answer()? - Kai Militzer
[Asterisk-Users] Callgroup with bristuff ISDN? - Massimo De Nadal
[Asterisk-Users] Announcement to caller when called party haspicked up - without initial Answer()? - Bob Goddard
[Asterisk-Users] SIP x NAT - Michael Giagnocavo
[Asterisk-Users] SPA-841 Call Waiting - Paul Dugas
[Asterisk-Users] Asterisk and Cisco phones chan_sccp vs chan_skinny vs native SIP and one-way audio - Michael J. Tubby B.Sc (Hons) G8TIC
[Asterisk-Users] Sending forwarded calls out to a different provider - Calvin Hendryx-Parker
[Asterisk-Users] Record inbound and outbound calls to and from one number. - bret...@worldcall.net
[Asterisk-Users] SIP x NAT - Charles S. Antrim
[Asterisk-Users] Return call after transfer with no answer - Christopher Slaght
[Asterisk-Users] Where does a newbie get started? - John Williams
[Asterisk-Users] PRI not hanging up the channel after Executing Hangup when dialing busy number. - James Sizemore
[Asterisk-Users] Multiport Fax over softphone - denon
[Asterisk-Users] chan_sccp bug / problem - Remco Barende
[Asterisk-Users] Q: PRI leading 0 (area access code) or 00 (country access code) missing on incoming callerid - Peter Svensson
[Asterisk-Users] Outlook Integration - Manjit Riat
[Asterisk-Users] PRI Dropped Calls - Audit, Restore, Idle state - Jeb Campbell
[Asterisk-Users] Delayed echo - Bart Roos
[Asterisk-Users] SIP x NAT - Michael Giagnocavo
[Asterisk-Users] Strange sip address? - Dan Zhou
[Asterisk-Users] Callerid on blind transfer w/ Cisco 7960 - Adam Fineberg
[Asterisk-Users] PRI Dropped Calls - Audit, Restore, Idle state - Peter Svensson
[Asterisk-Users] Cisco phones config over internet - Gregory Junker
[Asterisk-Users] Cisco phones config over internet - David Newman
[Asterisk-Users] A neat "hot seating" mplementation - Howard Lowndes
[Asterisk-Users] video conferencing bounty - dean collins
[Asterisk-Users] Developing an IP Phone - Mitchel Constantin
[Asterisk-Users] Cisco phones config over internet - Mitchel Constantin
[Asterisk-Users] Cisco 7960 and AutoAnswer. - C F
[Asterisk-Users] Basic Answering Machine Function? - Adam Goryachev
[Asterisk-Users] Grandstream stops working after "Register - Wolfgang S. Rupprecht
[Asterisk-Users] Asterisk with Grandstream ringback - Craig Guy
[Asterisk-Users] Budgetone ringing volume - James Doherty
[Asterisk-Users] RE: Answering Machine Function? - John Williams
[Asterisk-Users] RE: Answering Machine Function? - Wolfgang S. Rupprecht
[Asterisk-Users] Audio Quality over LAN very bad - Chamberland-Larose, Guillaume
[Asterisk-Users] Zap channels in AU hanging up on STD pips - Stuart Elvish
[Asterisk-Users] Telephone Line options in Asterisk - Daniel Wright
[Asterisk-Users] Intel chip IA98 - Max
[Asterisk-Users] Trunked IAX or not - Mark Eissler
[Asterisk-Users] RE: Answering Machine Function? - Glenn Powers
[Asterisk-Users] Asterisk and Cisco phones chan_sccp vs chan_skinny vs native SIP and one-way audio - Julien Goodwin
[Asterisk-Users] Timer for MeetMe on Mac OS X - Spencer Nassar
[Asterisk-Users] Audio Quality over LAN very bad - Nic le Roux

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