March 2007 Archives by thread
Starting: Thu Mar 1 00:06:10 MST 2007
Ending: Sat Mar 31 23:44:06 MST 2007
Messages: 2482
- [asterisk-users] extensions.conf & sccp.conf howto call external
number
Michiel van Baak
- [asterisk-users] read write or only read fields in cdr?
Bayrouni
- [asterisk-users] read write or only read fields in cdr?
Bayrouni
- [asterisk-users] DTMF not being detected with 1 provider. Works
with the other provider...
Evert
- [asterisk-users] Siemens HiPATH 3700 with Asterisk
Jorge de Diego
- [asterisk-users] How to get values of local channels context
Yuan LIU
- [asterisk-users] transfer function
Denis V. Gudtsov
- [asterisk-users] about bluetooth channel
Dave Cotton
- [asterisk-users] Not registering Port with VSP
Davy Chan
- [asterisk-users] FAX using T38
Zoa
- [asterisk-users] 1.4 lost internet internal
phones loose registration
Thomas Kenyon
- [asterisk-users] Re: queue information into db
Tomislav Parcina
- [asterisk-users] Re: fax support
Tomislav Parcina
- [asterisk-users] Re: SIP interface status and calllimit
Tomislav Parcina
- [asterisk-users] Issue with Calling Name ID in SIP: Asterisk sets
Caller ID Number as Name if NO Name
Jean-Marc Salsa
- [asterisk-users] TDM400p Loaded only once
Il Neofita
- [asterisk-users] Revolution Call Accounting Desktop
Tomislav Parcina
- [asterisk-users] voicemail advanced options problem with mysql
datbase
srinivas Antarvedi
- [asterisk-users] UK SIP Gateway
--
- [asterisk-users] 3 way calling independent of phone hw.
Simon Tennant
- [asterisk-users] Help Needed: Can't make "local" calls on a brand
new PRI
Matt
- [asterisk-users] Cannot hear ringback music from telco
Vincent Tam
- [asterisk-users] IAX best practices
Asterisk
- [Asterisk-Users] SIP to IAX - forcing codec pass thru
Thomas Kenyon
- [asterisk-users] Zaptel 1.4.0
Mike Hammett
- [asterisk-users] Snom 320 password
Mike Hammett
- [asterisk-users] Paid support offered
asterisk at 911networks.com
- [asterisk-users] blieve i my TE110P or My teleco provider ??
younss azzayani
- [asterisk-users] Zaptel 1.4.0
Ricardo Carvalho
- [asterisk-users] Testing asterisk with sipp
John Albano
- [asterisk-users] Extensions +International
Rob Schall
- [asterisk-users] Help: CallerID Name not being sent on
outboundPRI trunk
Webster, Andrew
- [asterisk-users] Call connected,
cannot hear or speak - $20 for fix
Supa
- [asterisk-users] Multiple simultaneous calls
stefano.totaro at transport.alstom.com
- [asterisk-users] About queues and multiple lines.
Nuria Fernandez
- [asterisk-users] Asterisk Realtime
Mike Hammett
- [asterisk-users] Fwd: Problem with TE212P
Benito Camelas
- [asterisk-users] Problem with TE212P
younss azzayani
- [asterisk-users] Help understanding SIP SHOW CHANNELS
Mojo with Horan & Company, LLC
- [asterisk-users] Tesco Internet Phone
Julian Lyndon-Smith
- [asterisk-users] Asterisk Realtime
Mike Hammett
- [asterisk-users] Voice mail is not giving unavailable or busy
prompts
Stephen Bosch
- [asterisk-users] Asterisk 1.4.1
Forrest Beck
- [asterisk-users] build rpm fails
Devraj Mukherjee
- [asterisk-users] Digium S101i - pickupexten doesn't work
Joseph
- [asterisk-users] gtalktovoip and Asteirsk
Klaverstyn, David C
- [asterisk-users] Polycom reject button
Jason Walker
- [asterisk-users] 2 Call locations
Jason Walker
- [asterisk-users] Help: CallerID Name not being sent on outbound
PRI trunk
C F
- [asterisk-users] RE: Polycom reject button
JR Richardson
- [asterisk-users] No Caller ID Name PRI NI2
C F
- [asterisk-users] How can I use the "GET VARIABLE variablename" in
AGI
李君
- [asterisk-users] Test
Wai Wu
- [asterisk-users] Newbie extensions.conf question
Chris Griffin
- [asterisk-users] Newbie extensions.conf question
Chris Griffin
- [asterisk-users] Multiple simultaneous calls
stefano.totaro at transport.alstom.com
- Rif: Re: [asterisk-users] Multiple simultaneous calls
stefano.totaro at transport.alstom.com
- [asterisk-users] BLF not working with Asterisk 1.4.0
Ricardo Carvalho
- [asterisk-users] Running Fax on E1 line
younss azzayani
- [asterisk-users] Re: Sending SMS
Tomislav Parcina
- [asterisk-users] Re: Asterisk and DTMF
Carlos Barros
- [asterisk-users] Cannot hear ringback music from telco
Vincent Tam
- [asterisk-users] Help Needed: Can't make "local" calls on a brand
new PRI
Andrew Latham
- [asterisk-users] cmd page crashes Asterisk SVN-branch-1.4-r57207
Hall, Eric M.
- [asterisk-users] T38
Khaled
- [asterisk-users] Problem with TE212P
Benito Camelas
- [asterisk-users] Multiple simultaneous calls
Stefano Totaro
- [asterisk-users] IAX best practices
Asterisk
- [asterisk-users] Asterisk and Fax
--[ UxBoD ]--
- [asterisk-users] Alec Saunders post about Mashable Telco's
Dean Collins
- [asterisk-users] DTMF detection problems on PRI channels?
Tony Mountifield
- [asterisk-users] WMI from Asterisk to Cisco Call Manager
Frédéric Marti
- [asterisk-users] IAX best practices
Asterisk
- [asterisk-users] Help Voicemail to SMS using asterisk
Goke Aruna
- [asterisk-users] Voicemail to SMS using asterisk
Goke Aruna
- [asterisk-users] Double DTMF digits sent on IAX native bridge
Remi Quezada
- [asterisk-users] DTMF from TDM400P and X100P
Yuan LIU
- FW: [asterisk-users] Alec Saunders post about Mashable Telco's
Dean Collins
- [asterisk-users] 1.4 - SLA
Russell Bryant
- [asterisk-users] Alec Saunders post about Mashable Telco's
Cory Andrews
- [asterisk-users] PRI progress codes.
John Bittner
- [asterisk-users] rtsavesysname not working in 1.4
David Thomas
- [asterisk-users] multiple phones registered for the same user
Andrew Joakimsen
- [asterisk-users] REMOTE CRASH FIX
Mike Lynchfield
- [asterisk-users] svn 1.4 - mp3 support and changing the
installation directory
tzieleniewski
- [asterisk-users] IP addresses
Mike Hammett
- [asterisk-users] How to log VERBOSE statement to a file?
Larry Alkoff
- [asterisk-users] Zaptel 1.2.15 Released
Asterisk Development Team
- [asterisk-users] Asterisk 1.2.16 Released
Asterisk Development Team
- [asterisk-users] Asterisk 1.4.1 Released
Asterisk Development Team
- [asterisk-users]
Need comparison between PBXtra, Trixbox, Thirdlane,
Druid, Aheeva etc.
Zeeshan Zakaria
- [asterisk-users] How to fail an AGI
Yuan LIU
- [asterisk-users] Asterisk - e164 (enum) lookup confused
Joseph
- [asterisk-users] Re: What means: Request to schedule in the past?!?!
Martin Joseph
- [asterisk-users] Somebody can help me?
Andrea Seghezzi
- [asterisk-users] gtalk2voip and Asterisk
Mani Sridhar
- [asterisk-users] hanging an asterisk box off of a PBX analog
extension
Mike D'Ambrogia
- [asterisk-users] creating new asterisk application
Giedrius Augys
- [asterisk-users] dial question
Hall, Eric M.
- [asterisk-users] Help: CallerID Name not being sent on outbound
PRI trunk
C F
- [asterisk-users] So does 1.4.1 show up in the /branches/1.4,
or only in the tags/1.4.1
Brad Templeton
- [asterisk-users] Voice quality issues
Alan Chandler
- [asterisk-users] running error: load_modules: No 'modules.conf'
found - vesrion 1.4.1 from svn
tzieleniewski
- [asterisk-users] When does local leg in call file start?
Yuan LIU
- [asterisk-users] x100p.com
Mail list
- [asterisk-users] Real Time, sip.conf, [general]
André Santos
- [asterisk-users] running error,
unable to load *.conf files: load_modules: No 'modules.conf'
found - svn version 1.4.1
TZieleniewski
- [asterisk-users] Configurations Files of TE110P
younss azzayani
- [asterisk-users] Is the 1.0.X branch vulnerable to the SIP issue?
Thermal Wetland
- [asterisk-users] Read() status?
Yuan LIU
- [asterisk-users] new kernel and zaptel
Giedrius Augys
- [asterisk-users] Re: Problem with TE212P
Benito Camelas
- [asterisk-users] HITBSecConf2007 - Malaysia: Call for Papers now
Open
Praburaajan
- [asterisk-users] HITBSecConf2007 - Malaysia: Call for Papers now
Open
Praburaajan
- [asterisk-users] SMS ON ASTERISK
Assis, Eduardo
- [asterisk-users] CTI
Nuria Fernandez
- [asterisk-users] Sangoma A102d and Asterisk on Debian 3.1.
Markus Monka
- [asterisk-users] TC400B
Wai Wu
- [asterisk-users] How to disable MOH completely?
David Thomas
- [asterisk-users] TDM400P/FXS in a HP DL380 G5
James FitzGibbon
- [asterisk-users] Rx+,Rx-,Tx+,Tx- of TE110P
younss azzayani
- [asterisk-users] Re: Asterisk Java w/ Threads
Stefan Reuter
- [asterisk-users] g.729 on solaris10/x86
Juraj Bednar
- [asterisk-users] Setting Sip Headers From Dial App?
Stuart Sheldon
- [asterisk-users] Voicemail question
Hall, Eric M.
- [asterisk-users] Using Asterisk as Voicemail Server on a dinosaur
Meridian System
J French
- [asterisk-users] [Announce] Web-MeetMe V3.0.1 released
Dan Austin
- [asterisk-users] extra-sounds 1.4.5 timestapmed newer than 1.4.6 ???
Mr. James W. Laferriere
- [asterisk-users] Polycom Questions
»Steven Ringwald«
- [asterisk-users] server generated outbound conference calls?
Dean Collins
- [asterisk-users] IAX2, DTMF and x86_64.
William F. Acker WB2FLW +1-303-722-7209
- [asterisk-users] app_queue not using exit context?
Steve Edwards
- [asterisk-users] Re: Registrations, how many is too many?
Tomislav Parcina
- [asterisk-users] Re: OT: Patch to OSS app for CDP without a Cisco
switch - TESTERS WANTED
Tomislav Parcina
- [asterisk-users] preventing voicemail pickup after SIP redirect ?
Louis-David Mitterrand
- [asterisk-users] Re: Asterisk Faxing Support
Tomislav Parcina
- [asterisk-users] Re: Digium cards on Vmware
Tomislav Parcina
- [asterisk-users] web based sipphone
Pezhman Lali
- [asterisk-users] Micros-Fidelio - billing in hotel
Tomislav Parcina
- [asterisk-users] [asterisk_voip] asterisk and ogg files
Bayrouni
- [asterisk-users] Digium cards on Vmware
Morten Isaksen
- [asterisk-users] visdn, misdn and the hell
Massimo Nuvoli
- [asterisk-users] Ringing does not terminate on mISDN after pickup
Arik Raffael Funke
- [asterisk-users] A New Phone Service - www.virtualphoneline.com
rehan at supertec.com
- [asterisk-users] Polycom 501 - Auto answer on one line appearance
Chris Mason (Lists)
- [asterisk-users] Asterisk 1.4.0 Installation error on Red Hat Linux
9.0-Urgent
Asterisk Asterisk
- [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850
mazhiyong at sohu.com
- [asterisk-users] Pickup failover
René Enskat
- [asterisk-users] chan_cellphone won't pair with phone
Earle Clubb
- [asterisk-users] Linksys PAP2 and Caller ID
Gergo Csibra
- [asterisk-users] Re:Problem with TE212P
Benito Camelas
- [asterisk-users] Long term voicemail archival and synchronisation
between multiple storage locations?
Dean Collins
- [asterisk-users] Building a new voicemail system... Testers needed!
Olle E Johansson
- [asterisk-users] running asterisk through cellphone
Mojo with Horan & Company, LLC
- [asterisk-users] How many gsm channels
Wai Wu
- [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'
Ken Williams
- [asterisk-users] Manager.conf '127.0.0.1 unable to authenticate'
Ken Williams
- [asterisk-users] Compiling smsq in 1.2
Yuan LIU
- [asterisk-users] Cancelling a digit in IVR
Kuba
- [asterisk-users] Add current caller to junk-callers-database
Alvin Austin
- [asterisk-users] Long term voicemail archival and
synchronisationbetween multiple storage locations?
Steve Totaro
- [asterisk-users] GTalk/Jabber passing audio in 1.4.1!
Ronald Lewis
- [asterisk-users] Nomination for Coolest App in 2007
Steve Totaro
- [asterisk-users] Anybody having problems using sellvoip?
Tom Lynn
- [asterisk-users] RE: Linksys PAP2 and Caller ID
Stewart Nelson
- [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 21
Justin Newman
- [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx - Call
droping issue
Vidura Senadeera
- [asterisk-users] Back to back E1 - asterisk <=> toshiba pbx -
Calldroping issue
Steve Totaro
- [asterisk-users] Asterisk 1.4.1 - Calling problem
Thomas Deillon
- [asterisk-users] capi installation problem
Giedrius Augys
- [asterisk-users] VoIP over Alvarion Wireless
Matt
- [asterisk-users] Background / Invalid Extension through cell phone
John Congdon
- [asterisk-users] TC400B
Wai Wu
- [asterisk-users] Realtime Extensions and "Include"
Peder at NetworkOblivion
- [asterisk-users] asterisk and ssl
nik600
- [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)
Steve Totaro
- [asterisk-users] Background / Invalid Extension through cell phone
Steve Totaro
- [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)
Steve Totaro
- [asterisk-users] asterisk and ssl
Steve Totaro
- [asterisk-users] queue information in mySQL
Thomas Winter
- [asterisk-users] Asterisk queue and agents
Hall, Eric M.
- [asterisk-users] Asterisk Auto-dial out
Phil Menico
- [asterisk-users] AsteriskNow Beta 4 - zaptel.conf / zapata.conf
problems
mark.d.rowe at bt.com
- [asterisk-users] queue information in mySQL
Steve Totaro
- [asterisk-users] mobility with asterisk
Alvaro Pacho
- [asterisk-users] Problem HandyTone 488 does not call transfer
Pablo Almido
- [asterisk-users] gtalk2voip and Asteris
DID 4ME
- [asterisk-users] Asterisk Registering to other SIP servers.
Bala Neelakantan
- [asterisk-users] Asterisk Realtime
Mike Hammett
- [asterisk-users] Re: Asterisk Realtime
Mike Hammett
- [asterisk-users] auto dialer
Hall, Eric M.
- [asterisk-users] Problems with Zaptel Drivers
Lutgring, Sam
- [asterisk-users] sip show channels
Bill Gibbs
- [asterisk-users] Number of SIP messages per minute
Mark Davies
- [asterisk-users] SIP remote crash bug
voiplist
- [asterisk-users] Re: Back to back E1 - asterisk <=> toshiba pbx -
Call droping
Vidura Senadeera
- [asterisk-users] Call recording and archiving
Voip Asterisk
- [Asterisk-Users] Re: Pickup *8 with CallerID
Olivier
- [asterisk-users] Timeouts not working
Richard Trenchard
- [asterisk-users] How to handle SIP-Callerid?
Andreas Anderson
- [asterisk-users] Queue Announcements for Operators
scott
- [asterisk-users] "pritimer" parameter in zapata.conf
Vidura Senadeera
- [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.
Henry Cobb
- [asterisk-users] Re: queue information into db
nik600
- [asterisk-users] Asterisk Auto-dial out
Phil Menico
- [asterisk-users] Hinting and Realtime
René Enskat
- [asterisk-users] Asterisk + Panasonic pbx
Sanspareils Greenlans
- [asterisk-users] New Linksys SPA Daylight Saving Time Rule for
US/Canada
Trevor G. Hammonds
- [asterisk-users] Fwd: Back to back E1 - asterisk <=> toshiba pbx
-Call droping
Steve Totaro
- [asterisk-users] AMI Originate and release channels
Paulo Vicentini
- [asterisk-users] Queue Announcements for Operators
Steve Totaro
- [asterisk-users] cmd pickup Problem
jroesch at bsradio.de
- [asterisk-users] Asterisk distributed deployment
ggonzalez at telviso.com.ar
- [asterisk-users] Re: Asterisk Realtime
Mike Hammett
- [asterisk-users] Accessing Voicemail by dialing own number
Chris Carey
- [asterisk-users] Queue announcing hold sequence instead of hold time
Drew Gibson
- [asterisk-users] Packet2Packet Bridging Questions
Daryl Jurbala
- [asterisk-users] Re: Asterisk distributed deployment
ggonzalez at telviso.com.ar
- [asterisk-users] Re: Asterisk distributed deployment
Steve Totaro
- [asterisk-users] Sender phone ringing while recipient talking
Nathan Bell
- [asterisk-users] transfers and CDR
Rodrigo Gonzalez
- [asterisk-users] Sender phone ringing while recipient talking
Steve Totaro
- [asterisk-users] Number of groups?
Webster, Andrew
- [asterisk-users] outdial to phone for new VM notification
end1r
- [asterisk-users] Empty Wildcard TDM400P as a MeetMe timer.
Steve Totaro
- [asterisk-users] No application 'Prefix' for extension in1.2x,
what app I have to use instead?
Rafael J. Risco G.V.
- [asterisk-users] Asterisk SIP to MAX TNT Gateway, Sporadic Echo
JR Richardson
- [asterisk-users] Zap Channel Deadlocks
Ron McCarthy
- [asterisk-users] Call load balancing
David Ruggles
- [asterisk-users] 1.4 compile issue
Wai Wu
- [asterisk-users] Re: Coaching in asterisk
Justin Newman
- [asterisk-users] Sender phone ringing while recipient talking
Steve Totaro
- [asterisk-users] Newbie Question
Chris Nighswonger
- [asterisk-users] OT Vonage V-Phone Adapter (Possible Hack)
Steve Totaro
- [asterisk-users] Is Allison going to be banned from foreign travel
over polar bears?
Steve Prior
- [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same
machine
Jose Bertuzzi
- [asterisk-users] Issues with a Linksys SPA 2102 and asterisk
Erick Perez
- [asterisk-users] Which VoIP router and switch to use for medium
size business
Zeeshan Zakaria
- [asterisk-users] Fwd: Can't hear any sound
Asterisk Asterisk
- [asterisk-users] Zaptel problem after upgrading to 1.2.16
Mark Davies
- [asterisk-users] When to use Echo Cancellation cards?
Zeeshan Zakaria
- [asterisk-users] Can't hear any sound (This time in plain text)
Asterisk Asterisk
- [asterisk-users] How to enter bridge_native_loop???
Santosh Raghuram
- [asterisk-users] RE: Zaptel problem after upgrading to 1.2.16
Mark Davies
- [asterisk-users] Is there any variable for Voicemail Password in
Asterisk
jamshed zaidi
- [asterisk-users] Re: Back to back E1 - asterisk <=> toshiba pbx -
Call droping issue [ ref:00D36mPe.50032wycQ:ref ]
Vidura Senadeera
- [asterisk-users] disable client side hangup after dialing 911
Patrick Fortin
- [asterisk-users] sip tunnel
Pezhman Lali
- [asterisk-users] YAACID and manager.conf security
Todd H
- [asterisk-users] Another Faxing Question
Rob Schall
- [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same
Jose Bertuzzi
- [asterisk-users] RE: Coaching in asterisk
Wai Wu
- [asterisk-users] RE: Coaching in asterisk
Wai Wu
- [asterisk-users] Which hylafax client ?
Olivier
- [asterisk-users] AEL #include file
Philipp Kempgen
- [asterisk-users] Cdr_mysql compile question
David Ruggles
- [asterisk-users] OS X Frequent console disconnects 1.4.1
Bruce Ferrell
- [asterisk-users] play file and action only stop if one defined key
has been pressed
Thomas Winter
- [asterisk-users] How to enter bridge_native_loop???
Santosh Raghuram
- [asterisk-users] Recorded file processing app wanted
Steve Edwards
- [asterisk-users] Polycom call parking feature and Asterisk call
parking
Stephen Bosch
- [asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN
Davis Sylvester III
- [asterisk-users] spandsp,
app_rxfax: apps_Makefile.patch v1.2 > v1.4 = No Workie!
Doug
- [asterisk-users] How to best manage my dial plans as the continue
to grow, and grow, and grow....
Christopher Aloi
- [asterisk-users] DTMF issue with TDM404
Al
- [asterisk-users] installation pb on debian etch
Philippe MONROUX
- [asterisk-users] Connecting two asterisk server.
Sanspareils Greenlans
- [asterisk-users] FastAGI Question
Lee Jenkins
- [asterisk-users] chan_misdn and Asterisk 1.4.1 - Early B3 not
working any more
Gary Hawkins
- [asterisk-users] asterisk on mini-itx
Mail Lists
- [asterisk-users] Polycom call parking feature and
Asteriskcallparking
Bill Gibbs
- [asterisk-users] Security and DTMF
Lee Jenkins
- [asterisk-users] how to determine the remote IP address when
dialing in ooh323
VoIP User
- [asterisk-users] IAX2 audio issues
Vulpes Velox
- [Asterisk-Users] Cannot get back chan_zap.so module!??
OCOSA List Acct.
- [asterisk-users] Re: Recorded file processing app wanted
Justin Newman
- [asterisk-users] Follow Up on Cannot get back chan_zap.so module!??
OCOSA List Acct.
- [asterisk-users] How to best manage my dial plans as the
continueto grow, and grow, and grow....
Steve Totaro
- [asterisk-users] Re: Recorded file processing app wanted
Steve Totaro
- [asterisk-users] Re: Recorded file processing app wanted
Steve Totaro
- [asterisk-users] Fix for TZ values updates for DST
Darrick Hartman
- [asterisk-users] How to enter bridge_native_loop???
Santosh Raghuram
- [asterisk-users] Recorded file processing app wanted
Steve Totaro
- [asterisk-users] Complicated callback solution
marcelobiz at comcast.net
- [asterisk-users] g711 -> iLBC garbled voice in 1.4?
Ray Jackson
- [asterisk-users] DST changes for the US
C F
- [asterisk-users] Re-parking (or transfer) a parked call
Barry D. Hassler
- [asterisk-users] Complicated callback solution
marcelobiz at comcast.net
- [asterisk-users] Asterisk and Databases
marcelobiz at comcast.net
- [asterisk-users] Problem configuring voice conference
Asterisk Asterisk
- [asterisk-users] Problems with Voice conferencing
John covici
- [asterisk-users] How many outgoing phone line/voip account do I
need?
Kurt Kuo
- [asterisk-users] Pickup group
Khaled Chehab
- [asterisk-users] Coming events in Europe
Olle E Johansson
- [asterisk-users] Problem configuring voice conference
Asterisk Asterisk
- [asterisk-users] Re: Problems with Voice conferencing
John covici
- [asterisk-users] _ALERT_INFO replacement in 1.4?
Nikhil Jogia
- [asterisk-users] Re: Help: CallerID Name not being sent
Matthew Warren
- [asterisk-users] Problem with H323
Sebastian Bozioreanu
- [asterisk-users] AMI - DBPut
Tomislav Parcina
- [asterisk-users] Citel Handset Gateway DST fix - FYI
Steven
- [asterisk-users] Rebooting ALL polycom phones
Mike
- [asterisk-users] Shoutcast music-on-hold
Jon Schøpzinsky
- [asterisk-users] Single sign on PC + phone?
Patrick
- [asterisk-users] New to Asterisk
NetSys Admin
- [asterisk-users] DST changes for the US
Darryl Dunkin
- [asterisk-users]
In Asterisk 1.4.x, Why Digium has two H323 Channels
Thiago Maluf
- [asterisk-users] Rebooting all Aastra phones
Matt
- [asterisk-users] DST 2007 Config for Cisco 7970
Gary T. Giesen
- [asterisk-users] Filter IDENT(Port 113) on Linksys router puts
remote extensions to one way audio
Zeeshan Zakaria
- [asterisk-users] ACM question
David Ruggles
- [asterisk-users] SIP unicode support ?
Klaus Darilion
- [asterisk-users] GXP-2000 DST Change
Ken Williams
- [asterisk-users] Create meetme conference rooms on the flight.
Wai Wu
- [asterisk-users] FW: Seamless Multi Office Asterisk Deployment
Brandon Comouche
- [asterisk-users] OT: Sipura DST Rules
Dave Fullerton
- [asterisk-users] LIDB/CNAM STORAGE DATABASE NEEDED
Matt
- [asterisk-users] Incase anyone wanted it - SNOM USA DST settings
Andrew Latham
- [asterisk-users] GXV3000 & Speakphone
Davis Sylvester III
- [asterisk-users] deprecated ALERT_INFO var andAMI's Originate
command
Octavio Ruiz (Ta^3)
- [asterisk-users] Polycom: warble on registration?
Ken D'Ambrosio
- [asterisk-users] New to Asterisk
NetSys Admin
- [asterisk-users] Call Back
Luis Claudio Santos
- [asterisk-users] Voicemails with occasional speeded up portions
Anthony Rodgers
- [asterisk-users] TDM-400, Polycom SIP phones, and echo problems
Stephen Bosch
- [asterisk-users] Playback 5% Too Fast?
David Brazier
- [asterisk-users] RE: Playback 0.5% Too Fast?
David Brazier
- [asterisk-users] great problem with sounds and ztdummy
Germán Aracil Boned
- [asterisk-users] Update Asterisk 1.2.12 to 1.4.1 ?
Noc Phibee
- [asterisk-users] MusicOnHold stops after upgrade from 1.4.0 to 1.4.1
Damian Adamski
- [asterisk-users] CDR and CallerID
Mike
- [asterisk-users] Queue - Call delivery problems - Asterisk 1.4 -
autofill=yes
equis software
- [asterisk-users] voicemail scenario
richard Coco
- [asterisk-users] French PRI channel - exact signaling used
Cedric MILLET
- [asterisk-users] IAX2 Question (Asterisk 1.4 tarball)
David Ruggles
- [asterisk-users] SIP hardphones with good jitter tolerance
Chris Bagnall
- [asterisk-users] DST and VM timestamp
Damon Estep
- [asterisk-users] 120 concurrent ZAP connections in asterisk open
edition. Is that possible?
Héctor Maldonado
- [asterisk-users] How to match wild card inside a GoToIf?
Ricardo Carvalho
- [asterisk-users] cisco sip firmware update for cisco 7970
Connolly, Tim
- [asterisk-users] Getting 7970 to update
Connolly, Tim
- [asterisk-users] How to match wild card inside a GoToIf?
Ken Williams
- [asterisk-users] How to match wild card inside a GoToIf?
Ken Williams
- [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 48
David Cook
- [asterisk-users] asterisk 1.2.15 fax
Khaled Chehab
- [asterisk-users] RE: In Asterisk 1.4.x,
Why Digium has two H323 channels?
Thiago Maluf
- [asterisk-users] Asterisknow with video and X-Lite not quite working
Benedikt Franz
- [asterisk-users] Press quotes needed from ENUM users on Asterisk
John Todd
- [asterisk-users] Digium S101i - Adapter DTMF works perfeclty
Joseph
- [asterisk-users] Re: Asterisknow with video and X-Lite not quite
working
dave cantera
- [asterisk-users] T1 Integrator Birch
John Schmerold
- [asterisk-users] strange things on call transfer
Christoph Fürstaller
- [asterisk-users] TDM-400, Polycom SIP phones, and echo problems
Peter Doyle
- [asterisk-users] beronet BN4S0
Thomas Stein
- [asterisk-users] ChanSpy with Record() : Doesnt seem to work for an
unbridged SIP call
Onglipo Chengoli
- [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING
nivlekch
- [asterisk-users] What happend to voip-info?
Nir Simionovich
- [asterisk-users] Earliest dial tone, after boot up.
joe acquisto
- [asterisk-users] Earliest dial tone, after boot up.
joe a.
- [asterisk-users] What is the best phone to get when using a headset?
Gareth Blades
- [asterisk-users] Autoprovisioning ST2030S
Christoph Fürstaller
- [asterisk-users] What happend to voip-info?
Steve Totaro
- [asterisk-users] Manager connection problems
Jordan Novak
- [asterisk-users] T1 Integrator Birch
Steve Totaro
- [asterisk-users] Zaptel version for asterisk 1.2.16
Wilson Pickett
- [asterisk-users] Nomination for Coolest App in 2007
Steve Totaro
- [asterisk-users] What is the best phone to get when using a
headset?
Steve Totaro
- [asterisk-users] what happened to asterisk wiki???
Rizwan Hisham
- [asterisk-users] Manager connection problems
Steve Totaro
- [asterisk-users] Manager connection problems
Steve Totaro
- [asterisk-users] [G.729] Input Gain
Victor Mateevitsi
- [asterisk-users] IVR after hangup
Benny Amorsen
- [asterisk-users] what happened to asterisk wiki???
Steve Totaro
- [asterisk-users] ${EXTEN} is limited to 17 characters under IAX ?
Oded Arbel
- [asterisk-users] Call center manager for Asterisk (Release 0.3)
nik600
- [asterisk-users] Can I use an Intertel IPPhone Plus 7704500 with
Asterisk somehow
Bill Chmura
- [asterisk-users] Call center manager for Asterisk (Release 0.3)
Steve Totaro
- [asterisk-users] IAX2 - Congestion
Mario Mayerle Filho
- [asterisk-users] Which SIP method/option to display a short text
message ?
Olivier
- [asterisk-users] Sped up recordings with 1.4.1
Joshua Thompson
- [asterisk-users] RE: what happened to asterisk wiki???
JR Richardson
- [asterisk-users] Packetization Rate
Matt
- [asterisk-users] what happened to asterisk wiki???
Steve Totaro
- [asterisk-users] While the VoIP-Info.org site is down...
Steve Totaro
- [asterisk-users] Patton 1400
Jean-Louis curty
- [asterisk-users] voip-info.org status update
James H Thompson
- [asterisk-users] Inbound PSTN CLID irratic with A200
Wireless
- [asterisk-users] Cisco 7912
Matt Putnam
- [asterisk-users] While the VoIP-Info.org site is down...
Steve Totaro
- [asterisk-users] Voip-Wiki Site Information
Matt
- [asterisk-users] DECT to SIP gateway experiences
Daniel Pittman
- [asterisk-users] DNIS/DNID
Mark Quitoriano
- [asterisk-users] AgentCallBackLogin Help!
Ron McCarthy
- [asterisk-users] Cost of Branded Equipment for Voip Provider
Implementation
ecleofe7 at yahoo.com
- [asterisk-users] MP3Player
Dominik Zalewski
- [asterisk-users] Meetme variables
Rizwan Hisham
- [asterisk-users] busy/hangup/answer detection in PRI E1 channels
Vidura Senadeera
- [asterisk-users] voip-info.org status update
Joe Greco
- [asterisk-users] snom led not working with asterisk 1.4.1
Giorgio Incantalupo
- [asterisk-users] qozap: t3 timer expired for span ...
Chris Earle (CBL)
- [asterisk-users] asterisk n-way call problem
Rizwan Hisham
- [asterisk-users] voip-info.org status update
Joe Greco
- [asterisk-users] Freepbx Incoming call's configuration
younss azzayani
- [asterisk-users] Dropped calls in Asterisk - A general question
J. Oquendo
- [asterisk-users] shutdown
josanchez at unicauca.edu.co
- [asterisk-users] A200 card problem
Todd H
- [asterisk-users] Incoming Caller ID
Rob Vinson
- [asterisk-users] sip_nat.conf - Asterisk with two Ethernet
Interfaces
Anjul Srivastava
- [asterisk-users] Re: zapata with Tiger3XX compilation error
pedro noticioso
- [asterisk-users] voip-info.org is back!
Sean Bright
- [asterisk-users] Help! Echo problem even at T1 PRI?
Vincent Tam
- [asterisk-users] voip-info.org status update
Gary Eck
- [asterisk-users] Re: busy/hangup/answer detection in PRI E1
channels (Vidura Senadeera)
Vidura Senadeera
- [asterisk-users] Voicechanger update for asterisk 1.4
Andreas Anderson
- [asterisk-users] Dell poweredge 860 acceptable for office
environment ?
Olivier
- [asterisk-users] SIP phone supporting more than 10 extension with a
call transfer command
younss azzayani
- [asterisk-users] Warning LSP Low
Rajeev Natarajan
- [asterisk-users] Transfer feature not working on asterisk 1.4.0
Rizwan Hisham
- [asterisk-users] transfer=mediaonly : can't hear nothing
Simone Cittadini
- [asterisk-users] Cepstral voices
Julian Lyndon-Smith
- [asterisk-users] Only secretary can call the boss, all others
only reach the secretary when dial the boss extension
Ricardo Carvalho
- [asterisk-users] asterisk 1.2 branch revision 53132 failed to
compile
Wilson Pickett
- [asterisk-users] Cisco + Asterisk list anyone?
Curt Shaffer
- [asterisk-users] MAX TNT Question
JR Richardson
- [asterisk-users] proposal: a new mailing list for asterisk 1.4,
why not?
Giorgio Incantalupo
- [asterisk-users] Re: Zaptel version for asterisk 1.2.16
Brent Torrenga
- [asterisk-users] Asterisk 1.2.13 Caller ID problem
Igor Shmukler
- [asterisk-users] FW: Microsoft buys Tellme
Dean Collins
- [asterisk-users] Error compiling zaptel 1.4.0
Chris Nighswonger
- [asterisk-users] Refund from SellVoip?
Tom Lynn
- [asterisk-users] DISA and repeating calls
Kuba
- [asterisk-users] What happend to voip-info?
Ken Williams
- [asterisk-users] Problems with MFCR2 and Meridian
Arturo Ochoa
- [asterisk-users] Pickup some else's call
Rob Schall
- [asterisk-users] Jajah.com like script?
Ritesh Agrawal
- [asterisk-users] Follow me on multiple numbers..
Ritesh Agrawal
- [asterisk-users] Channel stuck problem..
Ritesh Agrawal
- [asterisk-users] Monitor queue calls with Local channel agents
Miguel Paolino
- [asterisk-users] Follow me on multiple numbers..
Philippe Lindheimer
- [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 67
Yeruult Dorjsuren
- [asterisk-users] SMS Integration and SMS commands
Ignacio Ortega A.
- [asterisk-users] Call counter for sip misbehaving
Rizwan Hisham
- [asterisk-users] detecting missing end-of-queue records
Lenz
- [asterisk-users] Queues
Steve Kennedy
- [asterisk-users] 1.4 sample postgresql configs
Derek Whitten
- [asterisk-users] Conference server (or how to make a call with more
than 3 users)
Yehavi Bourvine +972-8-9489444
- [asterisk-users] Choppy sound with chan_capi + Fritz Card USB
asterisk-users at christophrothe.de
- [asterisk-users] Re: Replacement Wiki - options (Formerly 'status
of voip-info')
Davis Sylvester III
- [asterisk-users] T1 cable for Digium T1/E1 Cards
Jeronimo Romero
- [asterisk-users] camp on off-line phone
Leif Neland
- [asterisk-users] T1 cable for Digium T1/E1 Cards
Jeronimo Romero
- [asterisk-users] T1 cable for Digium T1/E1 Cards
Jeronimo Romero
- [asterisk-users] camp on off-line phone
Jeronimo Romero
- [asterisk-users] zttool always reports "OK" on TDM400P
Yuan LIU
- [asterisk-users] TE110P: Error ==> Asterisk died with code 1.
Jeronimo Romero
- [asterisk-users] Follow me on multiple numbers..
Philippe Lindheimer
- [asterisk-users] Conference server (or how to make a call with
more than 3 u
Yehavi Bourvine +972-8-9489444
- [asterisk-users] Conference server (or how to make a call
withmore than 3 u
Jon Schøpzinsky
- [asterisk-users] Conference server (or how to make a call
withmore than 3 u
Yehavi Bourvine +972-8-9489444
- [asterisk-users] Conference server (or how to make a
call withmorethan 3 u
Jon Schøpzinsky
- [asterisk-users] make calls with different phone numbers
younss azzayani
- [asterisk-users] TDM400p, no CLI activity
joe acquisto
- [asterisk-users] asterisk 1.4: choppy voicemail sound after upgrade
from 1.2.9.1
Giorgio Incantalupo
- [asterisk-users] T1 cable for Digium T1/E1 Cards
jacobso1
- [asterisk-users] no special context for sip peer
Christophorus Laube
- [asterisk-users] Conference server (or how to make a call with
more than 3 u
Yehavi Bourvine +972-8-9489444
- [asterisk-users] Queue App - Free agent and waiting calls
equis software
- [asterisk-users] Dial(Local/${EXTEN}@longdistance)?
Rizwan Hisham
- [asterisk-users] TDM400p, no CLI activity
joe a.
- [asterisk-users] Cepstral and numbers
Julian Lyndon-Smith
- [asterisk-users] One way dtmf tone on IAX
Wai Wu
- [asterisk-users] Configuring Faxs any help :)
younss azzayani
- [asterisk-users] Unicall Brazil
andreunimed
- [asterisk-users] Zaptel Dummy Driver
Darryl Dunkin
- [asterisk-users] GXP-2000 Phones with Cisco 3560 PoE Switch
Cory Andrews
- [asterisk-users] Zaptel Dummy Driver
Darryl Dunkin
- [asterisk-users] Teliax problems, they say use SIP,
more mature & better working than IAX
Scott Plante
- [asterisk-users] ExternalIVR() Dialplan function and Festival
David Ruggles
- [asterisk-users] H.323 with g729
Dovid B
- [asterisk-users] 1.4.1 - T38 Pass Through - Seeing some odd errors
but the fax works.....
Christopher Aloi
- [asterisk-users]
Festival works extension to extension, not on trunk
Chris Carey
- [asterisk-users] Microsoft launches first PABX
Dean Collins
- [asterisk-users] SIP provider did not send BYE if callee is an queue
Thomas Winter
- [asterisk-users] how to interconnection asterisk(sip) with mera
Julyono Kurniawan
- [asterisk-users] Which parameters of a live Asterisk server would
you monitor ?
Olivier
- [asterisk-users] Configuring Faxs any help :)
younss azzayani
- [asterisk-users] Zapateller not playing audio via SIP Trunk?
Benoit Panizzon
- [asterisk-users] PROGRESS code
Bill Gibbs
- [asterisk-users] error, install freePbx
Carlos Jerónimo
- [asterisk-users] Problem with "AT&T Maintenance" protocol in PRI
connection, no B+D channels available
Kanelbullar
- [asterisk-users] Configuring Faxs any help :)
Steve Totaro
- [asterisk-users] which spandsp for asterisk 1.2.16 (eom)
Wilson Pickett
- [asterisk-users] asterisk on debian
Josu Lazkano Lete
- [asterisk-users] modem passthru
Mark Farver
- [asterisk-users] Zaptel 1.2.16 Released
Asterisk Development Team
- [asterisk-users] Activating Incoming Demo
Eddie Johnson Jr
- [asterisk-users] Asterisk Automated Outbound Messaging
Cory Andrews
- [asterisk-users] High Pitched Noise
Rob Schall
- [asterisk-users] Can't Compile w/HPEC
Noah Miller
- [asterisk-users] codec_zap and Asterisk 1.4.1
Jeremiah Millay
- [asterisk-users] SIP/Polycom Issue, Asterisk 1.2.16, calls dropped
Timothy McKee
- [asterisk-users] wrong values in duration and billsec in CDR
Jovanny Saravia
- [asterisk-users] polycom random reboots
Louis-David Mitterrand
- [asterisk-users] polycom random reboots
joe a.
- [asterisk-users] Limit call duration
Suity Zsolt
- [asterisk-users] automated dialout detect forward
Mike Heininger
- [asterisk-users] About Pickup Grandstream
LKS GMAIL
- [asterisk-users] FWD outgoing problem
Bogdan GONCIULEA
- [Asterisk-Users] Sipura SPA-841 and firewall
Wilson Pickett
- [asterisk-users] Metaswitch help needed
Steve Edwards
- [asterisk-users] wct4xxp problem
Matija Turk
- [asterisk-users] How to get AEL2
Rizwan Hisham
- [asterisk-users] Asterisk 1.2.17 Released
Asterisk Development Team
- [asterisk-users] Asterisk 1.4.2 Released
Asterisk Development Team
- [asterisk-users] Asterisk with AudioCodes Mediant 2000
Kinjal Dixit
- [asterisk-users] reducing the number of extensions for every user
Rizwan Hisham
- [asterisk-users] FWD outgoing problem
Bogdan Gonciulea
- [asterisk-users] asterisk log analyzer
Eric Smith
- [asterisk-users] ses ActiveDirectory and also Ldap and Kerberos.
Carlos Jerónimo
- [asterisk-users] res_musiconhold.c:1243 load_module: No music on
hold classes configured
Thomas Winter
- [asterisk-users] PickUp a call with feature pickup (*8) from a IAX2
channel
Alvaro Parres
- [asterisk-users] Dlink i2eye
Bruce Reeves
- [asterisk-users] Cisco 7970 with skinny on * 1.4.1
Richard Klingler
- [asterisk-users] Asterisk 1.4.2 Requires Zaptel from 1.4 svn branch
for zap_chan?
Stuart Sheldon
- [asterisk-users] How to resolve CallerID from AudioCodes FXO
hind habaoui
- [asterisk-users] install and setup app_mp4 application
richard Coco
- [asterisk-users] Asterisk 1.4.2 chan_zap
Jeremiah Millay
- [asterisk-users] Voicemail mailbox number passed in connection?
Lutgring, Sam
- [asterisk-users] Looking for a terminations provider (carrier grade)
sip
- [asterisk-users] G729 'disappears' randomly
Rajeev Natarajan
- [asterisk-users] Too Many Open Files, Hung SIP Sessions,
Can I Increase File Count?
JR Richardson
- [asterisk-users] Cisco 30VIP Phone
Chris Nighswonger
- [asterisk-users] SIP peer disappearing
Luki
- [asterisk-users] CN=Diarmaid O'Loughlin/O=QAD1 is out of the office.
Diarmaid O'Loughlin
- [asterisk-users] A request for your input.
lmth at deakin.edu.au
- [asterisk-users] strange ring
Pryakhin Dimitry
- [asterisk-users] beronet BN8S0 and isdn phone
Thomas Stein
- [asterisk-users] fax and mISDN: a chimera?
Giorgio Incantalupo
- [asterisk-users] 302 Moved temporarely
Tomislav Parcina
- [asterisk-users] Test Message
Farooq Ahmed
- [asterisk-users] Digium b410p and 2.6.17 kernel bug?
Garth van Sittert
- [asterisk-users] Bridged ZAP calls do not release
Mitch Thompson
- [asterisk-users] Asterisk 1.4.2
Thomas Deillon
- [asterisk-users] ChanSpy and MeetMe
GDrayer at guesswho.com
- [asterisk-users] Gizmo project answers every call - can I use it in
hunt group?
Russell Horn
- [asterisk-users] using ael and extensions.conf togather?
Rizwan Hisham
- [asterisk-users] Asterisk x Mera MVTS
Hermann Wecke
- [asterisk-users] accepting a call, macros, and key presses.
Jason Wolfe
- [asterisk-users] hardware spec
phil.dawson at marnock.com
- [asterisk-users] chan_capi and only one B channel usable?
Torge Szczepanek
- [asterisk-users] Linksys/Sipura SPA-942 phones in larger deployments
Chris Bagnall
- [asterisk-users] managers
Todd H
- [asterisk-users] PASS CORRECT CALLER ID INFO TO OFFSITE TRANSFER
Davis Sylvester III
- [asterisk-users] SIP REFER ans Dialplan
kalle at esprITschmiede.de
- [asterisk-users] Problem in using Two BRi Cards in Asterisk
Farooq Ahmed
- [asterisk-users] About FAX and PAP2
Lukas
- [asterisk-users] Zaptel 1.4.1 Released
Asterisk Development Team
- [asterisk-users] Outbound SIP call from asterisk extension
Karthik Arumugam
- [asterisk-users] Microsoft launches first PABX
Anselm Martin Hoffmeister
- [asterisk-users] minimal asterisk for iax2 bridge
Francisco Pérez Botella
- [asterisk-users] ChanSpy and MeetMe
GDrayer at guesswho.com
- [asterisk-users] No Audio when integrating openSER and Asterisk ,
in NAT
raviprakash sunkara
- [asterisk-users] AsteriskNow Beta 4 with T1 Cards?
Brian K. Alexander,
Jr. (Vision Point Systems)
- [asterisk-users] AsteriskNow Beta 4 with T1 Cards?
Brian K. Alexander,
Jr. (Vision Point Systems)
- [asterisk-users] SRTP testers needed
marek cervenka
- [asterisk-users] no incoming dad with mISDN 1.1.1 and asterisk?
Louis-David Mitterrand
- [asterisk-users] cause 127
Thomas Stein
- [asterisk-users] Semi-OT: Use T.38 ATAs to Extend fax lines
Dave Fullerton
- [asterisk-users] Debian Asterisk and MeetMe
Alan Chandler
- [asterisk-users] SIP/IAX peers UNREACHABLE and audio loss
Edoardo Serra
- [asterisk-users] HUD Lite server on Debian
Giorgio Incantalupo
- [asterisk-users] Sendmail and exchange for voicemail integration
Jordan Novak
- [asterisk-users] Noob question regarding PCI 2.x & TDM400P Card
Barton Fisher
- [asterisk-users] PC / Phone Combo
Chris Gamble
- [asterisk-users] IAX2 certificate based (RSA) user auth in 1.4.x
Kai-Uwe Jensen
- [asterisk-users] Doorphone vs. Grandstream BT101
Jay Milk
- [asterisk-users] Problem with busy and unavailable
Stefan Guenther
- [asterisk-users] RE: PC / Phone Combo
Chris Gamble
- [asterisk-users] switches
phil.dawson at marnock.com
- [asterisk-users] SER vs Asterisk?
David Anderson
- [asterisk-users] freepbx -> DB Error messages...
Francesco Peeters (Asterisk)
- [asterisk-users] Issue with Hamlet ISDN PCI card(Cologne Chipset)
Mauro Zanin
- [asterisk-users] Asterisk Viruses?
Matthew Rubenstein
- [asterisk-users] Shoretel integration into Salesforce.com
Dean Collins
- [asterisk-users] Can be called on FreeWorldDialup/IAX channel,
but can't make calls
Timothy Parez
- [asterisk-users] Need feedback on vitelity
Mail list
- [asterisk-users] Voicemail for an AT&T System 75
Gary Eck
- [asterisk-users] TDM analog cards, volume, echo, fxotune,
ztmonitor and HPEC
Stephen Bosch
- [asterisk-users] Asterisk with Dialplan or TrixBox for this case?
Brian McEntire
- [asterisk-users] Remote host can't match request NOTIFY to call
Richard Klingler
- [asterisk-users] Doorphone vs. Grandstream BT101
marcotasto
- [asterisk-users] Question about DSP in Digium card
A. Levy
- [asterisk-users] Timeout for conferences
Andreas v. Heydwolff
- [asterisk-users] asterisk: error while loading shared libraries:
libiksemel.so.3:
cannot open shared object file: No such file or directory
Dmitri Smirnoff
- [asterisk-users] Problem with ztdummy
Alan Chandler
- [asterisk-users] Answer Confirmation with SIP/AIX channels
ASTERISKLIST at KEYWAYS.COM
- [asterisk-users] AOCD -> SendText()?
Andreas Anderson
- [asterisk-users] AOCD -> SendText()?
Andreas Anderson
- [asterisk-users] AOCD -> SendText()?
Andreas Anderson
- [asterisk-users] the age old telephone tree... why re-invent the
wheel?
dave cantera
- [asterisk-users] Anyone having trouble with claling US Domestic on
Sellvoip?
Salvatore Giudice
- [asterisk-users] voicemail is not playing messages
Joseph
- [asterisk-users] Answer Confirmation with SIP/AIX channels
Philippe Lindheimer
- [asterisk-users] FW: Seamless Multi Office Asterisk Deployment
kjcsb
- [asterisk-users] ztdummy install in the new zaptel 1.4.1
Dmitri Smirnoff
- [asterisk-users] how to check and set D-channel status
Farooq Ahmed
- [asterisk-users] Chan_cellphone and CentOS 4.x
Bruce Reeves
- [asterisk-users] Two or More Bri Cards
Farooq Ahmed
- [asterisk-users] Asterisk 1.4 Realtime problems
Callum McGillivray
- [asterisk-users] Failure acknowledgement time
cimsi
- [asterisk-users] Moving from Bristuff to mISDN
Olivier
- [asterisk-users] Failure acknowledgement time
cimsi
- [asterisk-users] cutting hash in dial app
René Enskat
- [asterisk-users] Counting callers
Suity Zsolt
- [asterisk-users] Asterisk incoming caller id problem
johnny_xing
- [asterisk-users] No Audio when integrating with openSER and
Asterisk in the SAME LAN ,
raviprakash sunkara
- [asterisk-users] ARI with * 1.4.x
Richard Klingler
- [asterisk-users] How is this feature called ?
Olivier
- [asterisk-users] Re: format_wav.c:247 update_header: Unable to
find our position
Bartek Bulzak
- [asterisk-users] Registration timed out after a "sip reload"
Gregory Duchatelet
- [asterisk-users] Asterisk and T38 ?
Noc Phibee
- [asterisk-users] Multi-registration ?
Olivier
- [asterisk-users] 1.4 - IAX2 - No registration for peer
dave cantera
- [asterisk-users] outbound call
Karthik Arumugam
- [asterisk-users] Polycom 601 loop
Nathan Bell
- Fwd: [asterisk-users] Multi-registration ?
Olivier
- [asterisk-users] Re: Polycom 601 loop
Nathan Bell
- [asterisk-users] SER vs Asterisk?
David Anderson
- [asterisk-users] Getting an ASR Number
Matt
- [asterisk-users] cutting hash in dial app
René Enskat
- [asterisk-users] Hang up detection time in FXS module
Gustavo Cordeiro
- [asterisk-users] Doorphone
Ray Wadkins
- [asterisk-users] SIP registration
Nathan Bell
- [asterisk-users] Device not registering after boot
Jay Moore
- [asterisk-users] Emergency chan_sip issue
Chris Bagnall
- [asterisk-users] SRTP vs ZRTP in Asterisk
Michael Graves
- [asterisk-users] SIP REFER
kalle.odenthal at genion.de
- [asterisk-users] Server Recomendation
Forrest Beck
- [asterisk-users] rx_fax and Asterisk 1.4.2
Heison Chak
- [asterisk-users] SRTP vs ZRTP in Asterisk
dubrowin.44628913 at bloglines.com
- [asterisk-users] SIP Video Camera
KokMengLoh
- [asterisk-users] how to define a pilot number
Lito Lampitoc
- [asterisk-users] Asterisk 1.4 and chan_misdn
Pierre Burton
- [asterisk-users] vzaphfc installation...
Mauro Zanin
- [asterisk-users] ztdummy and MOH
Klaverstyn, David C
- [asterisk-users] Re: how to define a pilot number
David Cook
- [asterisk-users] Using server side phonebook directory with SPA941
Maxim Veksler
- [asterisk-users] AMI - delete voicemail
Tomislav Parcina
- [asterisk-users] Asterisk MSOutlook Dialer
San Singhania
- [asterisk-users] UK BT PRI
Steve Kennedy
- [asterisk-users] cisco 7905
Khaled Chehab
- [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 106
David Cook
- [asterisk-users] IAX Experiences [WAS: Question about DSP in Digium
card]
Noah Miller
- [asterisk-users] AOC billing
Stefano Corsi
- [asterisk-users] TDM400p reliability
Joe Acquisto
- [asterisk-users] just call to user
Josu Lazkano Lete
- [asterisk-users] P-Asserted-Identify or Remote-Party-ID, or both?
Matt
- [asterisk-users] Park & No Announce?
Ken Williams
- [asterisk-users] Park & No Announce?
Ken Williams
- [asterisk-users] Park & No Announce?
Ken Williams
- [asterisk-users] Park & No Announce?
Ken Williams
- [asterisk-users] Erased log files
Luis Claudio Santos
- [asterisk-users] ARI with * 1.4.2 won't display recordings
Richard Klingler
- [asterisk-users] Inbound Voice Quality - Speed Change
Jim Duda
- [asterisk-users] "Couldn't load variables.txt?aldope=xxxxx "
Carlos Jerónimo
- [asterisk-users] Macro Dial - External DID
Forrest Beck
- [asterisk-users] just on my LAN
Josu Lazkano Lete
- [asterisk-users] Odd MeetMe bahaviour with MoH ...
Gordon Henderson
- [asterisk-users] Can I generate random SIP traffic?
gabriele.pinzauti at witech.it
- [asterisk-users] REG : H.323 Configurations with Asterisk
Anisha Kumar
- [asterisk-users] Voicemailmain not changing password?
Rizwan Hisham
- [asterisk-users] Trixbox 1.2.3 - TDM400 FXOs - Outgoing Calls
- Transfer # Not Wor king
José Luis Ledesma
- [asterisk-users] Friday asterisk users live conference/podcast at
12:30PM EDT
Wilson Pickett
- [asterisk-users] System from AMI
Tomislav Parcina
- [asterisk-users] Re: AOC billing
Tomislav Parcina
- [asterisk-users] wireless desktop phones
Jordan Novak
- [asterisk-users] * 1.4.1: connected to gtalk but no voice passing
Giorgio Incantalupo
- [asterisk-users] wireless desktop phones
Jordan Novak
- [asterisk-users] Meetme cant handle more than 5 users in a call??
hmmmm
Dean Collins
- [asterisk-users] MOS Score
Matt
- [asterisk-users] h323
Pezhman Lali
- [asterisk-users] Development of new features in Asterisk Manager
Moacir O. de Souza Junior - Personalsoft Sistemas Ltda.
- [asterisk-users] PoE - IEEE 802.3af
Mike
- [asterisk-users] Doorphone vs. Grandstream BT101
Steve Totaro
- [asterisk-users] Asterisk: recommended installation
Alejandro Cabrera Obed
- [asterisk-users] BRI Cards
Asterisk
- [asterisk-users] Wanted: German to English translator for Asterisk
documenation
Stefan Wintermeyer
- [asterisk-users] Transfering not working - how to debug?
Alan Chandler
- [asterisk-users] Multi-line phones - Asterisk uses wrong callerid
Drew Gibson
- [asterisk-users] App_RXFax Problem.
John Wulter
- [asterisk-users] SIP OPTIONS dialog not understood
Steve Edwards
- [asterisk-users] Polycom and Asterisk
Mike Hammett
- [asterisk-users] Unsetting Global Vars
Johann Hoehn
- [asterisk-users] Polycom and Asterisk
Darryl Dunkin
- [asterisk-users] asterisk-addons-1.4 write wrong uniqueid
Richard Klingler
- [asterisk-users] Call dies when I press *
Mike Diehl
- [asterisk-users] Polycom SoundPoint 501
Paolo Supino
- [asterisk-users] Polycom SoundPoint 501
Darryl Dunkin
- [asterisk-users] How to place a call to a Google Talk user?
Am Turnip
- [asterisk-users] Stepped deployment - T1 PRI passthru
Erik Anderson
- [asterisk-users] Nice Transfer Feature
Lacy Moore - Aspendora
- [asterisk-users] vzaphfc installation?
Mauro Zanin
- [asterisk-users] Voice mail
Khayelihle Mbona
- [asterisk-users] sip: failed the authenticate on INVITE
Michael Zoller
- [asterisk-users] Dell poweredge 860 acceptable
forofficeenvironment ?
joe a.
- [asterisk-users] Set(CALLERID(all) not working with 'unknown' call?
jan.sarin at securia.se
- [asterisk-users] error in FreePBX
Carlos Jerónimo
- [asterisk-users] cisco 7902
Khaled Chehab
- [asterisk-users] txfax and result
Giedrius Augys
- [asterisk-users] DISCONNECT 41 hangup problem on PRI
Tom De Wispelaere
- [asterisk-users] Nice Transfer Feature
Jeronimo Romero
- [asterisk-users] Interconnexion d'un serveur Asterisk à des PABX LG ( IP LDK)
khawla khawla
- [asterisk-users] Asterisk Feature attended transfer
Christoph Fürstaller
- SV: [asterisk-users] Set(CALLERID(all) not working with 'unknown'call?
jan.sarin at securia.se
- [asterisk-users] Off Topic: Open Source USB Softphone
Luis Claudio Santos
- [asterisk-users] Where are Spandsp changelogs or bugs available ?
Olivier
- [asterisk-users] L options in Dial() dont seem to work....
Mark Reardon
- [asterisk-users] maximum simultaneous calls
Mark Quitoriano
- [asterisk-users] Asterisk does not reINVITE after 302Redirect &
401Unauthorized
Mushtaq_Ahmed at 3com.com
- [asterisk-users] Is it possible to install CCM on a Linux platform ?
Olivier
- [asterisk-users] Scratchy Audio with Asterisk 1.2.4 over IAX on
FreeBSD?
Benoit Panizzon
- [asterisk-users] SIP & NAT
Mike Hammett
- [asterisk-users] Re: Problem converting a Cisco 7960 to SIP
Brad Stockdale
- [asterisk-users] help - UNSUBSCRIBE
Jerric
- [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits
kjcsb
- [asterisk-users] Hearing noise after 1min of calling
younss azzayani
- [asterisk-users] Polycom 501 + Asterisk +Edit buttons
Rob Schall
- [asterisk-users] bugetone 200's
phil.dawson at marnock.com
- [asterisk-users] Re: Problem converting a Cisco 7960 to SIP
Brad Stockdale
- [asterisk-users] chan_misdn
Tiziano Martelli
- [asterisk-users] UK PRI and outgoing CLI FYI
Steve Kennedy
- [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits
kjcsb
- [asterisk-users] DTMF Corruption Problem in 1.4.2 for SIP RFC2833
plz halp
Justin Tunney
- [asterisk-users] Call Waiting problems
Lachek Butalek
- [asterisk-users] error in FreePBX
Philippe Lindheimer
- [asterisk-users] Re: [asterisk-dev] Find the name of queue
Octavio Ruiz (Ta^3)
- [asterisk-users] Polycom Power
Mike Hammett
- [asterisk-users] CallerID + Name
Rob Schall
- [asterisk-users] Security on long distance calls
Stefano Corsi
- [asterisk-users] Setting rxgain per channel
Delca
- [asterisk-users] Linksys SPA 3102 causing me problems
Alan Chandler
- [asterisk-users] Problem while using asterisk Realtime
Sanjay Rajdev
- [asterisk-users] SIP RTP Tunnel
kalle.odenthal at genion.de
- [asterisk-users] Queue priority
Jordan Novak
- [asterisk-users] queue priority causes crash
Jordan Novak
- [asterisk-users] SIP RTP Tunnel
kalle.odenthal at genion.de
- [asterisk-users] SIP RTP Tunnel
kalle.odenthal at genion.de
- [asterisk-users] queue priority
Jordan Novak
- [asterisk-users] Need help to strip variable
van Leen, Phil
- FW: [asterisk-users] REG : H.323 Configurations with Asterisk
Anisha Kumar
- [asterisk-users] Correct latency values in "sip show peers"
Rolz
- [asterisk-users] Security on long distance calls
Rizwan Hisham
- [asterisk-users] Asterisk-Addon-1.4.0 MySQL
KC
- [asterisk-users] web based sip phone
Pezhman Lali
- [asterisk-users] bad case of buzzing
Louis-David Mitterrand
- [asterisk-users] Asterisk 1.4 with Digium B410P - Timing problem
Administrator TOOTAI
- [asterisk-users] Speed Dial Application in *
Chris Nighswonger
- [asterisk-users] Sipura SPA2000 Transfer Call
Chris Blunt
- [asterisk-users] RE: Polycom 501 + Asterisk +Edit buttons
JR Richardson
- [asterisk-users] Which IP Phones have buttons can be assigned to
functions with Asterisk
bilal ghayyad
- [asterisk-users] pickupgroup / SIP / Cisc phones
Michael Landin Hostbaek
- [asterisk-users] xten web phone
Pezhman Lali
- [asterisk-users] SPA3102 PSTN fallback
Todd H
- [asterisk-users] call file vs. originate
Nathan Bell
- [asterisk-users] One way intermittent static to PSTN
Porier, Jeremy M.
- [asterisk-users] SNOM 360
--[ UxBoD ]--
- [asterisk-users] unconditionally redirecting incoming calls by 302
Moved Temporarily messages doing right accounting
Ricardo Carvalho
- [asterisk-users] forwarding loop not detected
Bill Gibbs
- [asterisk-users] Multi-Level Queue
Peder at NetworkOblivion
- [asterisk-users] Realtime call-limit
Peder at NetworkOblivion
- [asterisk-users] Paging
Forrest Beck
- [asterisk-users] Redirect failed, channel not up.
Nathan Bell
- [asterisk-users] switchtype and signalling query
Simon Alman
- [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 128
fb at lucyde.com
- [asterisk-users] Re: Lucent TNT - ring timer
JR Richardson
- [asterisk-users] Indicating agent status on the phone
Alexander Topolanek
- [asterisk-users] Question on Priorities
--[ UxBoD ]--
- [asterisk-users] Meetme question
Adrian Marsh
- [asterisk-users] Sponsored development - Monodirectional audio
handling
Edoardo Serra
- [asterisk-users] wireless desktop phones
f6hqz-m at hamwlan.net
- [asterisk-users] LUSYN patches
Marc McLaughlin
- [asterisk-users] Re: Sponsored development - Monodirectional audio
handling
Edoardo Serra
- [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 129
fb at lucyde.com
- [asterisk-users] Understanding the dial flags
Alan Chandler
- [asterisk-users] Setting a call to be recorded before Xfer?
J French
- [asterisk-users] Re: Meetme question
Justin Hamade
Last message date:
Sat Mar 31 23:44:06 MST 2007
Archived on: Sat Mar 31 23:44:09 MST 2007
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