April 2009 Archives by thread
Starting: Wed Apr 1 00:39:05 CDT 2009
Ending: Thu Apr 30 23:34:51 CDT 2009
Messages: 1589
- [asterisk-users] DAHDI with OSLEC
Marco Sambo
- [asterisk-users] dynamic codec preferences
Steve Edwards
- [asterisk-users] What is the one thing that polycom can do...
Rob Hillis
- [asterisk-users] Queue data from within dialplan?
Lenz Emilitri
- [asterisk-users] What is the one thing that polycom can do...
randulo
- [asterisk-users] stress asterisk voicemail
Pepo
- [asterisk-users] codec payload size
Steve Underwood
- [asterisk-users] Remote host can't match request CANCEL to call
Shaun Wingrin
- [asterisk-users] Extract a MOS value from Asterisk CDR
Marc Leurent
- [asterisk-users] login-logout asterisk
Oguzhan Kayhan
- [asterisk-users] Trunk SIP and configuration
ludo perrot
- [asterisk-users] Asterisk doesn't relay remote MOH during hold
Anthony Plack
- [asterisk-users] Avoid compression with g.729/gsm/etc.
Anthony Plack
- [asterisk-users] [Zaptel] Why no driver for PCI voice modems?
Wilton Helm
- [asterisk-users] [Zaptel] Why no driver for PCI voice modems?
Tim Nelson
- [asterisk-users] [Zaptel] Why no driver for PCI voice modems?
Alan Lord (News)
- [asterisk-users] iax2 not registering at startup, works on reload
Yahya Mohammad
- [asterisk-users] SIP Context Confusion
Anthony Plack
- [asterisk-users] Asterisk 1.6.0.7 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk-Addons 1.6.2.0-beta1 Now Available
Asterisk Development Team
- [asterisk-users] SIP topology hiding
Martin
- [asterisk-users] 400 calls at g711 how much cpu power
Erick Perez
- [asterisk-users] Trying to test my voicemail
Pepo
- [asterisk-users] [CLOSED] Asterisk + OpenSIPs Integration - Rewrite URI on Trunk Numbers of a SIP Trunk
Marc Leurent
- [asterisk-users] PRI problem
Harry Vangberg
- [asterisk-users] [Closed] no ringtone - just silence/bridging ofexternal calls
alex.mosburger at orange-ftgroup.com
- [asterisk-users] async agi question
cyr2242 at gmail.com
- [asterisk-users] Friday April 3rd Gizmo, OpenSky, Skype for Asterisk, SIP for Skype - where are they?
randulo
- [asterisk-users] evaluate SIP response codes in dialplan
Marcus Hunger
- [asterisk-users] Mountain ahead of me!
Gabriel - IP Guys
- [asterisk-users] activate telco redirection service from Asterisk
Giorgio Incantalupo
- [asterisk-users] Asterisk SIP trunk to Cisco IAD2400
JR Richardson
- [asterisk-users] FXO Ignore ring
Cary Fitch
- [asterisk-users] fxotune and the bug
bilal ghayyad
- [asterisk-users] SIP vs RTP destination IP
David Ruggles
- [asterisk-users] Nothing at /proc/zaptel with new Digium TE201
criptos
- [asterisk-users] Asterisk + Cisco Call Manager
Timothy Smith
- [asterisk-users] opermode=?
bilal ghayyad
- [asterisk-users] Magic List: Thanks Shain Rufeel & Danny Nicholas.
criptos
- [asterisk-users] T1/PRI ignore answer signal
Jerry Geis
- [asterisk-users] meetme dahdi and zaptel
Dave Poirier
- [asterisk-users] VB6 to HUD Pro Integration
Gregory Malsack
- [asterisk-users] Asterisk G729 codec...
criptos
- [asterisk-users] Asterisk 1.2.32, 1.4.24.1, and 1.6.0.8 Now Available
Asterisk Development Team
- [asterisk-users] cant get a x100p works
Manolet Gmail
- [asterisk-users] problema con una x100p
Manolet Gmail
- [asterisk-users] 2-3 Calls at a time
David at ULC
- [asterisk-users] FXS Line Voltage When Dahdi/Zaptel is off?
Noah Miller
- [asterisk-users] Anyone actually built h323plus on Fedora?
sean darcy
- [asterisk-users] Simple Queue question
Haim Dimer
- [asterisk-users] Ring group howto
Michael
- [asterisk-users] Dahdi, TE220 Device, and Asterisk Problem
Elliot Murdock
- [asterisk-users] Dear asterisk-users at lists.digium.com USA Pharmacy ID 78634784
VIAGRA ® Official Site
- [asterisk-users] Asterisk and Call Manager
Timothy Smith
- [asterisk-users] VoIP Farm
Gabriel - IP Guys
- [asterisk-users] Local music on hold -- mohinterpret=passthrough assymetrical ?
Richard Brady
- [asterisk-users] agi no longer working with 1.4 svn 186229
John covici
- [asterisk-users] Unichan wtih Te201p alarms
criptos
- [asterisk-users] New ViciDial Call Center Suite Release: 2.0.5
Matt Florell
- [asterisk-users] Bridging Avaya IP systems and Cisco IP system
Gavin Henry
- [asterisk-users] Seg Fault after upgrade to Asterisk 1.6.0.8
M Hulber
- [asterisk-users] opermode=?
bilal ghayyad
- [asterisk-users] SIP Warnning Message
César García
- [asterisk-users] Using multiple 'peer' identities on one phone with 1.4
Florian Hackenberger
- [asterisk-users] Eicon Diva 2.01 PCI Passive BRI ISDN card
Puskás Zsolt
- [asterisk-users] conference calling
Danny Nicholas
- [asterisk-users] Grandstream surveillance devices
Jeff LaCoursiere
- [asterisk-users] Live Support function?
Dean Collins
- [asterisk-users] Live Support function?
Dean Collins
- [asterisk-users] TODAY April 4 -Global FSW Voice Meeting BerkeleyTIP -Linus, Guido, Shuttleworth...
john_re
- [asterisk-users] OT - Car Warranty Calls - OT
Cary Fitch
- [asterisk-users] Please help test the gender detection moduleat575-613-4392
Dovid Bender
- [asterisk-users] off topic - voip providers raided by FBI for unpaid telecom bills:
zoachien at securax.org
- [asterisk-users] Fwd: add a new queue strategy: SBR
Florian Hackenberger
- [asterisk-users] Inexpensive device for bandwidth management
Mike
- [asterisk-users] what can we do with lost voice packet on a congestioned VPN?
nik600
- [asterisk-users] Global h exten
Dovid Bender
- [asterisk-users] Global h exten
Dovid Bender
- [asterisk-users] Off-topic: SIP DTMF most supported method
Cesc Santa
- [asterisk-users] Fwd: add a new queue strategy: SBR
Andrey Solovjov
- [asterisk-users] app_queue.c: No one is answering queue
samuel
- [asterisk-users] fail to retrieve the calling party information
Rilawich Ango
- [asterisk-users] 25-50-100fxs
ContactTel Business
- [asterisk-users] SIP Registration and INVITE question
Steve Davies
- [asterisk-users] IPkall
Dean Collins
- [asterisk-users] OT - Call forwarding services for corporate users
Olivier
- [asterisk-users] Relay ringing sip message 180
Khaled W. Chehab
- [asterisk-users] Douds it
jibanez1971 at cimex.com.cu
- [asterisk-users] IOS Interface
Jorge Mendoza
- [asterisk-users] Provisioning GXP 2000
Philipp Kempgen
- [asterisk-users] IMAP Voicemail - can't get messages. Arrgh!
Noah Miller
- [asterisk-users] Sangoma and BT single lines
Ed W
- [asterisk-users] Asterisk 1.6.1.0-rc4 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk 1.6.0.9 Now Available
Asterisk Development Team
- [asterisk-users] Hacked
Jeremy Mann
- [asterisk-users] One way AUDIO
David at ULC
- [asterisk-users] Australian NBN network announced
Dean Collins
- [asterisk-users] asterisk and patton
mahboob zaman
- [asterisk-users] app_backticks and 1.6
Olivier
- [asterisk-users] OT - SIP MESSAGE, newline chars and formatting
Olivier
- [asterisk-users] Logging Asterisk console
Enrico Pasqualotto
- [asterisk-users] Zaptel connectivity issues
Danny Nicholas
- [asterisk-users] Grandstream blind transfer issue
Max Alex
- [asterisk-users] AEL2, BASE64_DECODE and hexadecimal
Olivier
- [asterisk-users] Best Practice Advice?
Gabriel - IP Guys
- [asterisk-users] is shared_lastcall available in 1.4
Gabriel Ortiz Lour
- [asterisk-users] Fwd: [asterisk-r2] MFCR2 With Unicall - can unicall affect zaptel commands?
Giovanny Magallanes
- [asterisk-users] i have a probleme and my asterisk and ovh
Henry
- [asterisk-users] dahdi_dummy: Unable to register DAHDI rtc driver
David Backeberg
- [asterisk-users] chan_mobile sms compatible phone
David fire
- [asterisk-users] Call Pickup Works w/Linksys ATA, not with Cisco 7940G
George Pajari
- [asterisk-users] Asterisk and Voice Recognition Sphinx
Marco Sambo
- [asterisk-users] Siemens Gigaset Phones get mute function.
Alan Lord (News)
- [asterisk-users] Asterisk Trunk billing
abdelkader
- [asterisk-users] Zopier Client
Gregory Malsack
- [asterisk-users] Perl AGI
michel freiha
- [asterisk-users] Multi Frequency Cycle Timeout - E1-R2 METROTEL COLOMBIA
Giovanny Magallanes
- [asterisk-users] Softphone question
David Ruggles
- [asterisk-users] Asterisk and WebIntegration
Kurian Thayil
- [asterisk-users] AstManProxy and broadcast
Olivier
- [asterisk-users] notifyringing=no does not work
Brad Finberg
- [asterisk-users] T.38 ATAs
Ian
- [asterisk-users] Check sip availability
criptos
- [asterisk-users] T.38 ATAs
Ian
- [asterisk-users] MeetMe not working - was before
John Rogers
- [asterisk-users] Looking for good IAX ATA
John Rogers
- [asterisk-users] DTMF
David at ULC
- [asterisk-users] IVR and DTMF
David at ULC
- [asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call Manager server?
Shocky
- [asterisk-users] Friday April 10th: Google Voice, Asterisk Open-Source Support, etc.
randulo
- [asterisk-users] Asterisk 1.4.24 and Gtalk audio failure
Administrator TOOTAI
- [asterisk-users] IVR Survey
James A. Shigley
- [asterisk-users] one-button call parking/pickup on Asterisk with Polycom phones?
Steve Johnson
- [asterisk-users] Looking for good IAX ATA
Giuseppe Barichello
- [asterisk-users] OT XEN asterisk and a digium board
David fire
- [asterisk-users] [astersik-users] ss7 consultancy $1000 USD
Apu Islam
- [asterisk-users] [OFF TOPIC] wich virtualization solution to use?
David fire
- [asterisk-users] asterisknow 1.5 with X100P and TDM400P
WipeOut
- [asterisk-users] problem with asterisk 1.4.24.1
troxlinux
- [asterisk-users] Asterisk is not designed for University with largeuser base?
Yehavi Bourvine
- [asterisk-users] Asterisk is not designed for University with large user base?
Nikolai Lusan
- [asterisk-users] Asterisk is not designed for University with largeuser base?
Atis Lezdins
- [asterisk-users] retransmision error con asterisk 1.4.24.1
troxlinux
- [asterisk-users] FAX reliability
Steve Underwood
- [asterisk-users] opensips and asterisk canreinvite
Nhadie
- [asterisk-users] Sending Re-Invite with Dialplan application?
Sai P. Varanasi
- [asterisk-users] Send Re-invite from Dialplan application?
Sai P. Varanasi
- [asterisk-users] Clock problem with TE122
Oguzhan Kayhan
- [asterisk-users] Agents on asterisk
"ROQUÉ, Francisco Emiliano"
- [asterisk-users] Agents on asterisk
"ROQUÉ, Francisco Emiliano"
- [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk
jonas kellens
- [asterisk-users] MySQL queries
Jeremy Mann
- [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk (update)
Dave Walker
- [asterisk-users] duration of rfc2833 generated dtmf
John covici
- [asterisk-users] wanpipe 3.2.7.1 Compiling error
Giovanni Andrés Nopal Pascual
- [asterisk-users] Changing menuselect values from CLI and not TUI
David Klaverstyn
- [asterisk-users] dynamic menus in dialplan
Eric Fort
- [asterisk-users] Ignoring time spent waiting in queue in CDR
Scott Gifford
- [asterisk-users] T.38 ATAs
Ian
- [asterisk-users] SIP and FW settings
Michael
- [asterisk-users] OT - Define what substitution is ...
Olivier
- [asterisk-users] Exit Dial Application
Christoph Fuerstaller
- [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
Florian Hackenberger
- [asterisk-users] .GSM -> .WAV (or ,MP3) Conversion
Tim Dobson
- [asterisk-users] SRTP testers needed
marek cervenka
- [asterisk-users] MOH
Khaled W. Chehab
- [asterisk-users] What means? Correct auth, but based on stale nonce received
Tiago Durante
- [asterisk-users] OT - snom phone question
Steve Davies
- [asterisk-users] Ring All Queue
Ryan M. Colbert
- [asterisk-users] FW: Asterisk-beginner : cannot make phone calls using Asterisk
Cary Fitch
- [asterisk-users] Asterisk Dial Pagers And Enter Callback Numbers
Supa
- [asterisk-users] Gxp 2000 softkey question
David Ruggles
- [asterisk-users] dial a pager and enter DTMF
Supa
- [asterisk-users] 2B Channel Transfer on XO-based T1
Max Metral
- [asterisk-users] RTCP ports
Michael
- [asterisk-users] astcanary not exiting in asterisk V1.6.1
Gerald Harshany
- [asterisk-users] What is "WARNING: Got 200 OK on REGISTER that isn't a register"?
Gerald Harshany
- [asterisk-users] Dear asterisk-users at lists.digium.com Pharmacy Online Sale 79% OFF!
VIAGRA ® Pfizer Inc.
- [asterisk-users] pickupexten *8
Gustavo A Gonzalez
- [asterisk-users] inbound filed
Bayardo Sanchez
- [asterisk-users] TDM2400P dial tone is not present on phones, but the phone ring with incoming calls
Giovanni Magallanes
- [asterisk-users] Friday Apr 17th @12 Noon ET: Digium's Open Source Asterisk Support
randulo
- [asterisk-users] reinvite problem
Oguzhan Kayhan
- [asterisk-users] Remote BLF / hint on IAX2 trunk
Marco Sambo
- [asterisk-users] ISDN from Macau CTM
Si Tai Fan
- [asterisk-users] AGI Programming
Alan Lord (News)
- [asterisk-users] mISDN ports and dstchannel CDR logging
Garth van Sittert
- [asterisk-users] How to send "404 Not found" SIP reply?
Chris Maciejewski
- [asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call
Vidura Senadeera
- [asterisk-users] Problem transferring calls between Cisco 7940 with SIP firmware
Massimiliano Stucchi
- [asterisk-users] Connection to non-human numbers
Danny Nicholas
- [asterisk-users] DTMF
Jeff LaCoursiere
- [asterisk-users] Simultaneous Calls at a time
David at ULC
- [asterisk-users] Set CDR(src) from dialplan
"ROQUÉ, Francisco Emiliano"
- [asterisk-users] AMI IAXPeers
Sebastian
- [asterisk-users] Sequential Ring Groups?
Marshall Henderson
- [asterisk-users] sending AT commands through the SIP channel to the end device?!
Tamer Higazi
- [asterisk-users] TDM2400P dial tone is not present on phones, but the phone ring with incoming calls
Giovanni Magallanes
- [asterisk-users] Stun clients and canreinvite
carl Lougher
- [asterisk-users] Canreinvite after media connection
carl Lougher
- [asterisk-users] how to call forward on 1.6
Oguzhan Kayhan
- [asterisk-users] async agi question
Jose Arias
- [asterisk-users] MOH always plays from the start
Mike
- [asterisk-users] Digium G.729 licenses
Arturo Díaz Almagro
- [asterisk-users] 1.4.21.1 - weird freeze
Mike
- [asterisk-users] 2BCT last mile... Hopefully
Max Metral
- [asterisk-users] Jabber and Presence
Gavin Henry
- [asterisk-users] Alcatel OmniPCX Enterprise + Asterisk with E1
Sebastian Milioto
- [asterisk-users] IAX IP Phone - PAP2Ts
bilal ghayyad
- [asterisk-users] Getting Zaptel and Asterisk Link, also Dahdi
bilal ghayyad
- [asterisk-users] opening 2 and more channels on 1 SIP account
Tamer Higazi
- [asterisk-users] Here is Step by Step Example of Asterisk PBX System Install and configuration
Dave Walker
- [asterisk-users] Sangoma A104d and Adtran 850 problems
Jim Dickenson
- [asterisk-users] Origination and Termination
Tom
- [asterisk-users] Callweaver TXfax queuing
Michael
- [asterisk-users] do i need to install libpri
Kashif Ali
- [asterisk-users] Digium Fax for Asterisk questions
Michael
- [asterisk-users] Insecure=
ContactTel Business
- [asterisk-users] Astlinux 0.6.5 upgrade released
Darrick Hartman
- [asterisk-users] asterisk-1.6.0.9-x86_64: voicemail: Segmentation fault (core dumped)
Justin Piszcz
- [asterisk-users] issue with sip 180 responses
Nir Levi
- [asterisk-users] Asterisk addons - disable H323
Michael
- [asterisk-users] Asterisk Addons - part 2
Michael
- [asterisk-users] Note for all regarding Asterisk Addons
Michael
- [asterisk-users] Asterisk 1.4 to 1.6 extensions.conf
Michael
- [asterisk-users] About Asterisk 1.6 web GUI
Gary Li
- [asterisk-users] Asterisk 'outgoing' directory
Michael
- [asterisk-users] T38 fax failing
Michael
- [asterisk-users] Zaptel to Dahdi
jonas kellens
- [asterisk-users] 1.4.24.1 freezing randomly - what files to use to downgrade
Mike
- [asterisk-users] Execute after hangup
Michael
- [asterisk-users] Voice mail does not contain a time?
Justin Piszcz
- [asterisk-users] Asterisk PA system with cepstral
Justin Killen
- [asterisk-users] Peer 'iaxfax' is now UNREACHABLE! Time: 3
Marco
- [asterisk-users] Asterisk process ended
Adrien Lemoine
- [asterisk-users] Should I go for Asterisk 1.6 ?
--[ UxBoD ]--
- [asterisk-users] Polycom wideband codecs?
mgraves at mstvp.com
- [asterisk-users] Asterisk process ended
Adrien Lemoine
- [asterisk-users] run dialplan when open line
michel freiha
- [asterisk-users] Polycom wideband codecs?
mgraves at mstvp.com
- [asterisk-users] Asterisk 1.6.1.0-rc5 Now Available
Asterisk Development Team
- [asterisk-users] Faxing and TIFF files
Michael
- [asterisk-users] Asterisk routine maintenance activities
James Mutuku
- [asterisk-users] CDR feature not working properly for "failed call attempt"
Vikas
- [asterisk-users] mISDN DTMF endless tone
Arturo Díaz Almagro
- [asterisk-users] E1 not synchronized
Anton Raharja
- [asterisk-users] Should you use UserEvents for monitoring calls ?
Olivier
- [asterisk-users] [asterisk-dev] How to get to 10.000 open calls
Atis Lezdins
- [asterisk-users] How to get to 10.000 open calls
Philipp Kempgen
- [asterisk-users] Zaptel tone debug
"ROQUÉ, Francisco Emiliano"
- [asterisk-users] Asterisk process ended
Adrien Lemoine
- [asterisk-users] Conference problem
Cristi Iconaru
- [asterisk-users] HI
Anil Kumar K
- [asterisk-users] Upgrading from 1.4.21.2 to 1.6.0.5 breaks sql queries with backslashes?
Kristina Harris
- [asterisk-users] how to know the channel from the iax phone side?
David fire
- [asterisk-users] random hangups: how to debug?
sean darcy
- [asterisk-users] voice quality
Rilawich Ango
- [asterisk-users] Conference problem
Cristi Iconaru
- [asterisk-users] Parked calls for multiple customers
carl Lougher
- [asterisk-users] Asterisk and HUD server
Marco Sambo
- [asterisk-users] Howto see the source ip address of SIP call in cli monitor
Shaun Wingrin
- [asterisk-users] Compact, fanless appliance?
Vincent
- [asterisk-users] Asterisk Capacity
Geraint Lee
- [asterisk-users] Cause 34 still there
Steve Davies
- [asterisk-users] UserEvent doc : is Uniqueid mandatory in 1.6
Olivier
- [asterisk-users] Asterisk Double Invite
Khaled W. Chehab
- [asterisk-users] Zaptel Not Releasing Channel (PRI)
Steve Totaro
- [asterisk-users] Do I need G729 codec for wholesale ?
Dusan Djordjevic
- [asterisk-users] Dial-out via AMI
Nhadie
- [asterisk-users] Fritz USB 2.1 on Asterisk 1.4.22 / trixbox
Akos Gabriel
- [asterisk-users] AMD Not Working
Sam Hawkin
- [asterisk-users] AGI PHP script
James A. Shigley
- [asterisk-users] CDR issue
Gustavo A Gonzalez
- [asterisk-users] Libpri-1.4.10 Released
Asterisk Development Team
- [asterisk-users] Convert file in GSM codec to G729 codec
Shaun Wingrin
- [asterisk-users] Record in mp3
Jose Enes Mateus
- [asterisk-users] dial and transfer while ringing
Olivier
- [asterisk-users] want to set up text based "adventure" for asterisk
Eric Fort
- [asterisk-users] CDR issue
Gustavo A Gonzalez
- [asterisk-users] BLINDTRANSFER and SIP hardphones
Olivier
- [asterisk-users] about Asterisk and AudioCodes FXO/H323
Huseyin Sahbal
- [asterisk-users] Dahi-tools Compilation on Ubuntu/Xen
info at ameri.me
- [asterisk-users] Hangup Detection After Originate (Asterisk Manager API)
Saurabh Nirkhey
- [asterisk-users] cheap CHEAP ata
David fire
- [asterisk-users] Feature request: manager show events
Olivier
- [asterisk-users] function originate
Rilawich Ango
- [asterisk-users] Friday Apr 24 @ 12 Noon: Wideband, or HD Voice as Polycom calls it
randulo
- [asterisk-users] timing source problem
Wolfgang Pichler
- [asterisk-users] FOP and UserEvent()
Marco Sambo
- [asterisk-users] listen to prompt before bridging call.
Deepak
- [asterisk-users] Duplicating existing PBX function
David Ruggles
- [asterisk-users] Record in mp3
Jose Enes Mateus
- [asterisk-users] Asterisk 1.6.2 Beta
Christian
- [asterisk-users] Dialtones as Progressinband
Timm M.Schneider
- [asterisk-users] Record in mp3
Jose Enes Mateus
- [asterisk-users] dahdi_tool reports that dahdi_dummy is UNCONFIGURED
David Backeberg
- [asterisk-users] voicemail number of rings
Adam Moffett
- [asterisk-users] Digium Fax for Asterisk
Anthony Cascante
- [asterisk-users] Asterisk & EC2
Aryan Ameri
- [asterisk-users] Can't dial out until I dial in once
Michael van der Stoop
- [asterisk-users] Callweaver/Asterisk 'outgoing' spool
Michael
- [asterisk-users] sip.conf RTP settings
Michael
- [asterisk-users] 64bit: any problems with asterisk?
sean darcy
- [asterisk-users] Outgoing Queues
Sebastian
- [asterisk-users] Digium fax failing
Michael
- [asterisk-users] FW: issue with sip 180 responses
Nir Levi
- [asterisk-users] Digium fax force T38?
Michael
- [asterisk-users] Digium fax failing
Michael
- [asterisk-users] 1.6.1: menuselect has problems with x86_64 ??
sean darcy
- [asterisk-users] Digium fax failing
Michael
- [asterisk-users] file.c:655 ast_openstream_full: File /tmp/winkel-gesloten.alaw does not exist in any format
jonas kellens
- [asterisk-users] sipgate doesn't work with sipgate anymore
Michael Obster
- [asterisk-users] 1.6.1: "DNS error" but ping works
sean darcy
- [asterisk-users] Error, Clue to what?
Cary Fitch
- [asterisk-users] Video Conference Software (Open Source)
joko pitoyo
- [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
--[ UxBoD ]--
- [asterisk-users] music on hold using mms
Rilawich Ango
- [asterisk-users] Going to AMOOCON?
randulo
- [asterisk-users] Diference between volume of mp3 and wav files
Jose Enes Mateus
- [asterisk-users] Change Termination of Read Command
Danny Nicholas
- [asterisk-users] SIP infrastructure
Philipp Kempgen
- [asterisk-users] Packet2packet bridging while in sip.conf canreinvite=no
jonas kellens
- [asterisk-users] IPv6 support?
Andrew Ruthven
- [asterisk-users] Where I get free VoiP-in numbers?
almidoster at gmail.com
- [asterisk-users] Who has the clever Polycom upgrade system?
Barry L. Kline
- [asterisk-users] POS modems
Steve Underwood
- [asterisk-users] finding the right amd.conf settings
Roi Stork
- [asterisk-users] Call recording - posible to remove recorded file at the end of the call
Christian Gansberger
- [asterisk-users] SIP CallerID Question
Geraint Lee
- [asterisk-users] no source on calllogs
Oguzhan Kayhan
- [asterisk-users] Explain when DIALSTATUS is set to CANCEL
Olivier
- [asterisk-users] How to get PBX's clock with AMI?
Daniel - Asterisk
- [asterisk-users] How to get PBX's clock with AMI?
Daniel - Asterisk
- [asterisk-users] Asterisk-Verifone-Agi
Juan Miguel Quiros Arrieta
- [asterisk-users] asterisk -C option not honored 100%
David Siefert
- [asterisk-users] Asterisk 1.6 and CDR/MySQL
--[ UxBoD ]--
- [asterisk-users] Asterisk-Addons 1.4.8 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk-Addons 1.6.1.0 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk 1.6.1.0 Now Available
Asterisk Development Team
- [asterisk-users] I am looking for a good source of Monterrey DIDs
Robert Augustyn
- [asterisk-users] Cisco SPA525G
Gondar Monn
- [asterisk-users] Replacement of Macro() with Gosub()
Steve Davies
- [asterisk-users] problem in upgrading to 1.6.1.0
Oguzhan Kayhan
- [asterisk-users] Diference between volume of mp3 and wav files
Jose Enes Mateus
- [asterisk-users] Diference between volume of mp3 and wav files
Jose Enes Mateus
- [asterisk-users] Something wrong with DAHDI signalling according to the CLI
jonas kellens
- [asterisk-users] Asterisk sudden crash
Andrew Nowrot
- [asterisk-users] Verifone-Asterisk-AGI
Steve Edwards
- [asterisk-users] Bounty for parking on <slot>@<context>
Steve Edwards
- [asterisk-users] What do I need to connect landline calls without telephony hardware?
don rhummy
- [asterisk-users] 2nd Parking Lot
Angvall
- [asterisk-users] ExtenSpy d option 1.6
Steve Casto
- [asterisk-users] automon *1 not working; asterisk-1.4.22.1
Joseph
- [asterisk-users] valetparking.c
Peder
- [asterisk-users] Asterisk and Shoretel integration
Andrea Brancatelli
- [asterisk-users] Asterisk or Zaptel Issues
Aman Dhally
- [asterisk-users] need help on asterisk call forwarding
Oguzhan Kayhan
- [asterisk-users] Saturday May 2 - Asterisk @ Global FSW Conference via VOIP - BerkeleyTIP - 21 Videos - For forwarding
john_re
- [asterisk-users] 2nd Parking Lot
Peder
- [asterisk-users] FW: Update: HD Communications Summit in NYC
Dean Collins
- [asterisk-users] rtsp help
Jerry Geis
- [asterisk-users] Question with Asterisk and call waiting ${CALLERID(name/num)}
Justin Piszcz
- [asterisk-users] ${CALLERID(name)} question
Justin Piszcz
- [asterisk-users] 2nd Parking Lot
Peder
- [asterisk-users] Wanpipe
Jeremy Mann
- [asterisk-users] Proper install order for Asterisk and it's related packages..
Jonathan Moore
- [asterisk-users] ExtenSpy d option 1.6
Steve Casto
- [asterisk-users] test
James A. Shigley
- [asterisk-users] Registration of 'cstore' rejected: 'Registration Refused' from: '62.213.196.38'
jonas kellens
- [asterisk-users] Voicemail Caller ID
admin at thelastestate.com
- [asterisk-users] What do I need to connect landline calls without telephony hardware?
Dave Walker
- [asterisk-users] What do I need to connect landline calls without telephony hardware?
Dave Walker
- [asterisk-users] Asterisk and 4G
Dave Walker
Last message date:
Thu Apr 30 23:34:51 CDT 2009
Archived on: Thu Apr 30 23:35:04 CDT 2009
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