August 2006 Archives by thread
Starting: Tue Aug 1 00:02:23 MST 2006
Ending: Thu Aug 31 23:46:24 MST 2006
Messages: 3082
- [asterisk-users] Re: question about asterisk DB
Tzafrir Cohen
- [asterisk-users] VoipNow 1.2.0 Beta
Michiel van Baak
- [asterisk-users] cmd DIAL - Who picked up the call?
Kai Ober
- [asterisk-users] SRTP help
Khaled Chehab
- [asterisk-users] Permission for files generated by voicemail
Jean-Yves Avenard
- [asterisk-users] asterisk gui
vivek at staff.ownmail.com
- [asterisk-users] SoftHangup with Polycom_acd_functions release of
asterisk
Dean at INKnBITs
- [asterisk-users] MWI from Asterisk to Meridian
Andrew Kohlsmith
- [asterisk-users] Re: Re: FYI - first release of alarm response code.
Steven
- [asterisk-users] nat and qualify questions
BerkHolz, Steven
- [asterisk-users] MWI from Asterisk to Meridian
Andrew Kohlsmith
- [asterisk-users] Is there a smarter way to ban expensive calls in
dial plan?
Chris Blunt
- [asterisk-users] AddQueueMember and Local channel
Asterisk
- [asterisk-users] Missing Fast AGI calling 'h' exten without hanging
up
Tony Mountifield
- [asterisk-users] Help debugging strange asterisk behaviour
jan.sarin at securia.se
- [asterisk-users] Media direct from IAX Phone to IAX Phone
Kamran Ahmad
- [asterisk-users] SV: Help debugging strange asterisk behaviour
jan.sarin at securia.se
- [asterisk-users] Problem with distortion of initial voicemail prompt
Frank Tarczynski
- [asterisk-users] Park / ParkAndAnnounce
Guillermo Roditi
- [asterisk-users] SendText() & displaying text messages
onaSIPhandset's screen
Guillermo Roditi
- [asterisk-users] Multi Asterisk Server to relay call request
Stephen Wingfield
- [asterisk-users] can't retake call after dialing through Zap/E1
wich doesn't answer
Manrique Feoli
- [asterisk-users] Extend analog phone via SIP (OT)
Ira
- [asterisk-users] IAX over two T1 connections bad quality
Tim Panton
- [asterisk-users] Controllable hold music
Thomas Kenyon
- [asterisk-users] nat and qualify questions
Alyed Tzompa
- [asterisk-users] Codec selection / IAX tunnels
Thomas Kenyon
- [asterisk-users] IAX and Accountcode
Douglas Garstang
- [asterisk-users] Re: How to configure NOKIA N70 with Asterisk?
Benny Amorsen
- [asterisk-users] MySQL 5.0+ and the MySQL addon - Can use stored
procedures?
Rushowr
- [asterisk-users] Line drops
J. Oquendo
- [asterisk-users] Re: Operator Console(s)/Shared Call Appearances
Mr. Jones
- [asterisk-users] IAX and Accountcode
Douglas Garstang
- [asterisk-users] AddQueueMember and Local channel
Asterisk
- [asterisk-users] ISDN incoming call - inband info and announcements
BEFORE ANSWER
Michal Dole ž el
- [asterisk-users] rx_fax problem
Paradise Dove
- [asterisk-users] Dundi and Dial Arguments
Mitch Sharp
- [asterisk-users] IAX and Accountcode
Douglas Garstang
- [asterisk-users] Unicall stack, right versions?
Barzilai
- [asterisk-users] Dundi and Dial Arguments
Douglas Garstang
- [asterisk-users] codec conversion
Wasif
- [asterisk-users] Dundi and Dial Arguments
Douglas Garstang
- [asterisk-users] Polycom IP600 HTTP Provisioning problem
VaibhaV Sharma
- [asterisk-users] Strange behaviour Panasonic KX-TD1232
Pablo Mora
- [asterisk-users] FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found
Zen Kato
- [asterisk-users] Asterisk with VoIP phone
J Rangi
- [asterisk-users] VOIP phone for Receptionist use
Jeff Busch
- [asterisk-users] RE: asterisk-users Digest, Vol 25, Issue 2
AstATN
- [asterisk-users] ANNOUNCE: libss7
Matthew Fredrickson
- [asterisk-users] Strange behaviour Panasonic KX-TD1232
Pablo Mora
- [asterisk-users] MWI from Asterisk to Meridian
C F
- [asterisk-users] A2Billing - destination
Luciano Moreira
- [asterisk-users] Re: AEL2 Looping
Steve Murphy
- [asterisk-users] softhangup() problem
Shaun Hofer
- [asterisk-users] cmd DIAL - Who picked up the call?
Eric "ManxPower" Wieling
- [asterisk-users] SER local as an Asterisk Trunk
Nhadie Ramos
- SV: [asterisk-users] VOIP phone for Receptionist use
Jon Schøpzinsky
- [asterisk-users] Problem with Cisco7970 SIP load / call transfer
Juha Suhonen
- [asterisk-users] Asterisk config with Analouge Audio Codec model
number MP108FXS
Mr shobhit nirala
- [asterisk-users] SRTP
harrygaillac-sip at yahoo.fr
- [asterisk-users] Slow dialing from PBX via E1
Gavin Hamill
- SV: [asterisk-users] Help debugging strange asterisk behaviour
jan.sarin at securia.se
- SV: [asterisk-users] Help debugging strange asterisk behaviour
jan.sarin at securia.se
- [asterisk-users] newbie - suggestions on installing Asterisk for
SOHO
Pele Zico
- [asterisk-users] Follow ON calling on DISA
sdcharly at gmail.com
- [asterisk-users] FXO module burn out !?
Rostislav Bagrov
- [asterisk-users] polycom soundstation 501 crash
Stas Khromoy
- [asterisk-users] Dell Poweredge 1950 / 2950
Frédéric Marti
- [asterisk-users] Call Routing based on Caller-Id
Matthew Crocker
- [asterisk-users] Playback() does not work
Camilo Echeverry
- [asterisk-users] Strange behaviour Panasonic KX-TD1232
Pablo Mora
- [asterisk-users] Asterisk, Linksys SPA-3000 echo
Dean at INKnBITs
- [asterisk-users] Arrays ???
Pele Zico
- [asterisk-users] Issue with IAX2 and Real Time configuration
Facundo Ameal
- [asterisk-users] [ANN] - Coder Needed for Patch
Bart Fisher
- [asterisk-users] SIP_HEADER() read-only
Vincent Regnard
- [asterisk-users] Limitations of IAX
Douglas Garstang
- [asterisk-users] canreinvite=yes and RTP dropping in and out
Gary Richardson
- [asterisk-users] DTMF intermittent on menu.
Shane Burrell
- [asterisk-users] asterisk optimizing
Jack Wei
- [asterisk-users] sip phone networking question [possibly OT]
Mojo with Horan & Company, LLC
- [asterisk-users] sip phone networking question [possibly OT]
Colin Anderson
- [asterisk-users] Strange behavior with SIP registration/connectivity
Ronald Lewis
- [asterisk-users] GSM analogue router
Garth van Sittert
- [asterisk-users] Rookie question, trying to learn
Randy Paries
- [asterisk-users] Limitations of IAX
Douglas Garstang
- [asterisk-users] IAX Trunking
Douglas Garstang
- [asterisk-users] Limitations of IAX
Douglas Garstang
- [asterisk-users] RemoveQueueMember isn't working.
Keith Herrington
- [asterisk-users] creidt card processing sripts for asterisk
Joseph
- [asterisk-users] DUNDi with SIP
Douglas Garstang
- [asterisk-users] DUNDi with SIP
Douglas Garstang
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] DUNDi with SIP
Douglas Garstang
- [asterisk-users] unsubscribe
Keith Herrington
- [asterisk-users] Ateus Easy gate call progress
Jan Fousek
- [asterisk-users] DUNDi with SIP
Douglas Garstang
- [asterisk-users] DUNDi with SIP
Douglas Garstang
- [asterisk-users] DUNDi with SIP
Douglas Garstang
- [asterisk-users] chan_zap.c: Failed to read gains: Invalid argument
jan.sarin at securia.se
- [asterisk-users] Re: need a pointer regarding scripting asterisk
Andy Kuo
- [asterisk-users] RE: Asterisk with VoIP phone (shadowym)
J Rangi
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] need dialout help in python script
shawn bright
- [asterisk-users] Re: DUNDi with SIP
Watkins, Bradley
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Re: DUNDi with SIP
Watkins, Bradley
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Re: DUNDi with SIP
Watkins, Bradley
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Clocking Multiple T1 Cards
Kevin P. Fleming
- [asterisk-users] create custom cdr's
Kevin P. Fleming
- [asterisk-users] Binary/unreadable configuration files?
Kevin P. Fleming
- [asterisk-users] Caller ID on Transfers
Kevin P. Fleming
- [asterisk-users] IP JitterBuffer for 1.2.5
Thierry Querette
- [asterisk-users] Re: DUNDi with SIP
Watkins, Bradley
- Subject: [asterisk-users] Slow dialing from PBX via E1
AstATN
- [asterisk-users] Asterisk 1.0.1 in SuSE 10.0
Oscar Carrillo
- [asterisk-users] About Digium cards and HP DL servers
Angel Gomez
- [Asterisk-Users] Duration for billing
Steve Edwards
- [asterisk-users] [OT] FYI: Polycom phone intermittent disconnects
Jeff Turner
- [asterisk-users] BRIDGEPEER and DIALEDPEERNAME empty
Koopmann, Jan-Peter
- [asterisk-users] IAX2 Trunking CPU usage
Jon Schøpzinsky
- [asterisk-users] Garbled initial voicemail prompt
Frank Tarczynski
- [asterisk-users] queue in realtime
unplug
- [asterisk-users] How to check if channel varaible have been set/not
empty?
Jan du Toit
- [asterisk-users] What I can use with ASTERISK to call clients to
remind them about their appointments
dmitri smirnoff
- [asterisk-users] volume adjustment?
Kohler, Jeffrey
- [asterisk-users] [OT] FYI: Polycom phone intermittent disconnects
Bill Gibbs
- [asterisk-users] Ringing all extensions
J. Oquendo
- [asterisk-users] wip 300 opensource code - changes to support SIP
MESSAGE
Jerry Geis
- [asterisk-users] MoH native volume
Carlos Chavez
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Asterisk H323 and Alcatel 4400
Carlos Chavez
- [asterisk-users] SIP_HEADER() read-only
Douglas Garstang
- [asterisk-users] Reboot Mediatrix
Julian Varanini
- [asterisk-users] IAX Trunking
Douglas Garstang
- [asterisk-users] VoiceMail being cutoff when leaving message
Randy Paries
- [asterisk-users] IAX Variables
Douglas Garstang
- [asterisk-users] IAX Variables
Douglas Garstang
- [asterisk-users] SIP_HEADER() read-only
Douglas Garstang
- [asterisk-users] Forbidden - wrong password on authentication for
INVITE
Brent Torrenga
- [asterisk-users] Queue bug: When 2 callers call in,
only one is processed until the first is answered
Keith Herrington
- [asterisk-users] Re: DUNDi with SIP
Watkins, Bradley
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] How to forward a call to an outside line
Dan Casey
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Detecting voicemail from CO on FXO port and
passing to H.323 phone. Possible?
Bob Bosiljevac
- [asterisk-users] Detecting voicemail from CO on FXO port
andpassing to H.323 phone. Possible?
Steven Totaro
- [asterisk-users] Detecting voicemail from CO on FXO port
Paul Davidson
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] About Digium cards and HP DL servers
angom at telnor.net
- [asterisk-users] About Digium cards and HP DL servers
VOICEIN at aol.com
- [asterisk-users] Re: DUNDi with SIP
Watkins, Bradley
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Using Flite in a call file.
Joey McDonald
- [asterisk-users] Run a script at certain CLI writes
Bart Fisher
- [asterisk-users] Echo cancell
Pablo Allietti
- [asterisk-users] Detecting voicemail from CO on FXO
Paul Davidson
- [asterisk-users] Strange behaviour Panasonic KX-TD1232
Pablo Mora
- [asterisk-users] Prevent a Polycom contact list to be overwritten
Stephen Murphy
- [asterisk-users] trinary expression
John Williams
- [asterisk-users] Encoding recorded queue calls to mp3
jan.sarin at securia.se
- [asterisk-users] Problem dialing out with a TDB400P
Dante Passalacqua
- [asterisk-users] Strange behaviour Panasonic KX-TD1232
Pablo Mora
- [asterisk-users] New UK prompts
Steve Kennedy
- [asterisk-users] Prevent a Polycom contact list to be overwritten
Douglas Garstang
- [asterisk-users] MWI from Asterisk to Meridian
AstATN
- [asterisk-users] Ringing all extensions
AstATN
- [asterisk-users] Opinions on Rhino PCI FXO cards
shadowym
- [asterisk-users] cannot received calls in pstn line
Lito Lampitoc
- [asterisk-users] Asterisk@Home, call reporting and performance
Esteban Guana-Jarrin
- [asterisk-users] asterisk dosenot compile
vivek at staff.ownmail.com
- [asterisk-users] Asterisk with AVM B1 and HFC
Marco Dieckhoff
- [asterisk-users] ANI agi
Sharon Lim
- [asterisk-users] Configuring meetme recording quality (8kHz to
32kHz or higher)
Jan du Toit
- [asterisk-users] How to connect Snom softphone from my home?
Crazy Boy
- SV: [asterisk-users] Help debugging strange asterisk behaviour
jan.sarin at securia.se
- [asterisk-users] Re: DUNDi with SIP
Watkins, Bradley
- [asterisk-users] Load balancing of IAX2
Kamran Ahmad
- [asterisk-users] creidt card processing sripts for asterisk
Dennis Nacino
- [asterisk-users] speech gaps with iax2
Pavel Jezek
- [asterisk-users] How to connect Snom softphone from my home?
Guido Hecken
- SV: [asterisk-users] Help debugging strange asterisk behaviour
(update)
jan.sarin at securia.se
- [Asterisk-Users] How to configure NOKIA N70 with Asterisk?
Crazy Boy
- [asterisk-users] sendtext() to another machine
Jerry Geis
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] SIP/Qualify
Dovid Bender
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Asterisk Manager Vb 6
Dovid Bender
- [asterisk-users] Sangoma A200 and Disconnected Cables
Andres
- [asterisk-users] SNOM 360
Dovid Bender
- [asterisk-users] SNOM 360
Dovid Bender
- [asterisk-users] SIP and podcasts
Dovid Bender
- [asterisk-users] (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems
Steve Totaro
- [asterisk-users] AgentCallBackLogin+Queue
Gleidson Antonio Henriques
- [asterisk-users] Re: DUNDi with SIP
Douglas Garstang
- [asterisk-users] Jabber questions
Julian Lyndon-Smith
- [asterisk-users] AgentCallBackLogin+Queue
Guido Hecken
- [asterisk-users] Festival Not Working
Jon Scottorn
- [asterisk-users] Is the manager good for high traffic?? but only
with one connection to it
Manrique Feoli
- [asterisk-users] Problems with monitor / mixmonitor stopping if
using Local channels
Julian Lyndon-Smith
- [asterisk-users] Dialplan routing based on CallerID
Matthew Crocker
- [asterisk-users] Steve Totaro I am trying to reach you.
Ferguson, Michael
- [asterisk-users] Running AGI in background
Bromont -
- [asterisk-users] Mediatrix 1204 and Asterisk 1.2.10
Julian Varanini
- [asterisk-users] DISA + Voicemail + DTMF
Dave
- [asterisk-users] Asterisk and Siemens Legacy PBX
James Arscott
- [asterisk-users] Simple config question
John Williams
- [asterisk-users] How to play music on hold from within PHP AGI
scripts?
Leo Burd
- [asterisk-users] Aastra VLAN issues
Kris Seraphine
- [asterisk-users] Setting CALLERID on a residential telco line
hugolivude
- [asterisk-users] Check call duration on active call in CLI?
voiplist
- [asterisk-users] autocreatepeer in iax
Kamran Ahmad
- [asterisk-users] Help - call recording being cut short if
transferred
Julian Lyndon-Smith
- [asterisk-users] cisco 2600
FaberK
- [asterisk-users] Fax tone detected, but no fax extension for CAPI
Stefan-Michael. Guenther (in-put GbR)
- [asterisk-users] [Solution] Call Asterisk from GoogleTalk and have
it tell you the status of your IAX2 links.
Matt Riddell (NZ)
- [asterisk-users] Japanese Sound Files
Nhadie
- [asterisk-users] load average with MOH
Jon Farmer
- [asterisk-users] Asterisk and Siemens Legacy PBX
James Arscott
- [asterisk-users] how to check the status of a channel
Thomas Artner
- [asterisk-users] Help with perl AGI script
Roy Kidder
- [asterisk-users] g729 and trafic
Walter Willis
- [asterisk-users] Linksys SPA-3000 Administration Guide
Marcos Rubino
- [asterisk-users] for some of my users,
VoiceMail is being cutoff when leaving message
Randy Paries
- [asterisk-users] Help with perl AGI script
Roy Kidder
- [asterisk-users] Using a DB for Configurations
Barry Fawthrop
- [asterisk-users] Asterisk and Siemens Legacy PBX
James Arscott
- [asterisk-users] previous reload of asterisk did not finish
John covici
- [asterisk-users] Ring Groups
Chris Hembrow
- [asterisk-users] AG-168V not registering.
sdcharly at gmail.com
- [asterisk-users] Anyone use TACI.pl for a click to call app? -
Doesn't seem to want to work for me
Christopher Aloi
- [asterisk-users] Variables sip redirects and call forward
C F
- [asterisk-users] How to "emulate" Music on Hold in a PHP AGI script?
Leo Burd
- [asterisk-users] Asterisk and Siemens Legacy PBX
AstATN
- [asterisk-users] HP ProLiant and Digium 24xxp
Robert Roach
- [asterisk-users] New people in this world and his problem with ISDN
Dominik Kiełb
- [asterisk-users] Video Conferencing over Asterisk
Siqhamo Sifo
- [asterisk-users] SER + Asterisk PSTN calls don't hung up
Ricardo Carvalho
- [asterisk-users] Ztdummy - No audio in BackGround function
support_list
- [asterisk-users] How to "emulate" Music on Hold in a PHP AGI
script?
Leo Burd
- [asterisk-users] Re: Load balancing of IAX2
Kamran Ahmad
- [asterisk-users] G729, IAX, polycom - trying to using 2 codecs
Dean at INKnBITs
- [asterisk-users] Inbound problems, no audio
Pablo Salvador Capo
- [asterisk-users] Caller ID problem on TDM400 FXO
Greg Delgado
- [asterisk-users] Re: [asterisk-dev] Questions regarding g.729 and
g.711 in Asterisk
Rich Adamson
- [asterisk-users] DTMF problems
Kohler, Jeffrey
- [asterisk-users] Ragi without rails possible ?
shawn bright
- [asterisk-users] Conditional branching
bails
- [asterisk-users] Re: [asterisk-dev] Tuning Software Echo Cancellers
Rich Adamson
- [asterisk-users] Hotels...
Ken D'Ambrosio
- [asterisk-users] Fwd: * and GTalk testing
David Freeman
- [asterisk-users] By week extension dialing
Dan Brummer
- [asterisk-users] looking to pay a consultant to help with my
asterisk installation
Randy Paries
- [asterisk-users] Hotels...
Jonathan k. Creasy
- [asterisk-users] E1 for Voice and Data with MFC/R2
Carlos Chavez
- [asterisk-users] RE: By week extension dialing
Brent Torrenga
- [asterisk-users] res_sqlite problems
Michael Iedema
- [asterisk-users] RE: By week extension dialing
Dan Brummer
- [asterisk-users] By week extension dialing
Dan Brummer
- [asterisk-users] "Off-circuits are busy now. Please try your call
again later"
Wolfgang Paul Rauchholz
- [asterisk-users] Re: By week extension dialing
Dan Brummer
- [asterisk-users] DTMF problems
Kohler, Jeffrey
- [asterisk-users] MOH Silence
Douglas Garstang
- [asterisk-users] sip incoming stop working,
what to look for in logs?
T. Shaw
- [asterisk-users] By week extension dialing
Dan Brummer
- [asterisk-users] FXS gateway/Channel Bank
Roger Workman
- [asterisk-users] FXS gateway/Channel Bank
Eric "ManxPower" Wieling
- [asterisk-users] Voicemail Platform
Roger Workman
- [asterisk-users] voicemail in mp3 format
Ever Zalazar
- [asterisk-users] Voicemail Platform
John Novack
- [asterisk-users] Voicemail Platform
Roger Workman
- [asterisk-users] Re: Meetme chat room with many users,
and only 4 can talk, is there a max amount of users?
Manrique Feoli
- [asterisk-users] SIP musicclass
Douglas Garstang
- [asterisk-users] NVFaxDetect and 1.2.10
John D. Coleman
- [asterisk-users] Music On Hold Class Not Makin' Sense
Douglas Garstang
- [asterisk-users] sip incoming stop working,
what to look for in logs?
T. Shaw
- [asterisk-users] PROBLEM MUSIC ON HOLD
Elpidio Ramos
- [asterisk-users] (no subject)
Sony Veri Shandy
- [asterisk-users] agi script runs even if no answer
shawn bright
- [asterisk-users] Polycom 301 and Linksys SRW224P PoE Switch
Santiago del Castillo
- [asterisk-users] VoipNow 1.2.0 Beta
Greg Boehnlein
- [asterisk-users] Polycom 1.6.7 Firmware Messages Button
Greg Boehnlein
- [asterisk-users] problem- 0:10 long message
ismir saljic
- [asterisk-users] Bluetooth phone as FXS/FXO with asterisk?
asterisk at anime.net
- [Asterisk-Users] ISDN Y cable
Olivier
- [asterisk-users] set minimum iax jitterbuffer
Pavel Jezek
- [asterisk-users] AGI doesn't execute PHP5 script
Stefan-Michael. Guenther (in-put GbR)
- [asterisk-users] Problems with Codecs in Asterisk
Chan Kwang Mien
- [asterisk-users] RE VoipNow 1.2.0 Beta
Matthew Warren
- [asterisk-users] codec_g729a.so coredump in SVN trunk
ast-lists at delvar.com
- [asterisk-users] IAX trunk behing NAT with dynamic IP
Andre Courchesne - Consultant
- SV: [asterisk-users] IAX trunk behing NAT with dynamic IP
Jon Schøpzinsky
- [asterisk-users] help with app_sms and chan_capi
Patrick Zwahlen
- [asterisk-users] Asterisk with BT's broadband voice service.
Ronan Mullally
- [asterisk-users] Jitterbuffer on SIP
Thierry Querette
- V: [asterisk-users] IAX trunk behing NAT with dynamic IP
Andre Courchesne - Consultant
- [asterisk-users] Problems with Codecs in Asterisk
Chan Kwang Mien
- [asterisk-users] Stopping Queue after nobody picked up the call . .
Ralph Liebessohn
- [asterisk-users] Probelm with IAX peers
David Freeman
- [asterisk-users] Sipura SPA-841
Giedrius Augys
- [asterisk-users] Cisco Phone Configuration Tool cannot find Files
Nate List
- [asterisk-users] Zaptel trunk failed to compile - Still but another
error
Administrator TOOTAI
- [asterisk-users] Asterisk and failover
Alex Lake
- [asterisk-users] A question about AGI and RECORD FILE
Randy Paries