May 2009 Archives by thread
Starting: Fri May 1 00:17:43 CDT 2009
Ending: Sun May 31 23:43:32 CDT 2009
Messages: 1486
- [asterisk-users] Cisco SPA525G
Gondar Monn
- [asterisk-users] What do I need to connect landline calls without telephony hardware?
don rhummy
- [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
--[ UxBoD ]--
- [asterisk-users] cdr_mysql custom fields ?
Kev Szaszvari
- [asterisk-users] Convert PC Audio wav file to g722
Sai P. Varanasi
- [asterisk-users] Asterisk and 4G
Gordon Henderson
- [asterisk-users] Failed log in
Cary Fitch
- [asterisk-users] May 1st @12 Noon: VoIP and home automation and control
randulo
- [asterisk-users] FW: Update: HD Communications Summit in NYC
randulo
- [asterisk-users] New system for recording - SCSI, SAS or SATA?
Tony Mountifield
- [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
--[ UxBoD ]--
- [asterisk-users] LoadAvg , Codec and Bandwidth Utilisation
David at ULC
- [asterisk-users] AMD Not Working
Sam Hawkin
- [asterisk-users] US Caller ID
Andrew Joakimsen
- [asterisk-users] Asterisk-Verifone-Agi
Andrew Joakimsen
- [asterisk-users] AGI - Ways to create a call
Tiago Durante
- [asterisk-users] 2 phone extensions on a single conference room
David at ULC
- [asterisk-users] Can someone help me with my IAX-registration
jonas kellens
- [asterisk-users] ISDN Error Code 42
Nitesh Divecha
- [asterisk-users] Asterisk and ODBC
Vela Sivasankaran
- [asterisk-users] Sangoma Wanpipe Driver Compile for DAHDI Failure
Atlanticnynex
- [asterisk-users] Sangoma Wanpipe Driver Compile for DAHDI Failure
Atlanticnynex
- [asterisk-users] LoadAvg , Codec and Bandwidth Utilisation
Dave Walker
- [asterisk-users] SIP Extension Registration and Security
Dave Walker
- [asterisk-users] Asterisk not starting up due to database problems
Charlie Grosvenor
- [asterisk-users] System noise reduction
Sunny Du
- [asterisk-users] asterisk-users Digest, Vol 58, Issue 5
Venefax
- [asterisk-users] Asterisk and ODBC
Kev Szaszvari
- [asterisk-users] question of flite installation
Rilawich Ango
- [asterisk-users] How to get PBX's clock with AMI?
Paul Hales
- [asterisk-users] How to get PBX's clock with AMI?
Paul Hales
- [asterisk-users] need help on asterisk call forwarding
Oguzhan Kayhan
- [asterisk-users] how to add applications to 1.6???
Oguzhan Kayhan
- [asterisk-users] Compact, fanless appliance?
Vincent
- [asterisk-users] advice on OrderlyStats (or other cc software)
Louis-David Mitterrand
- [asterisk-users] hint limitation
Antoine Patte
- [asterisk-users] Compact, fanless appliance?
Paulo Garcia
- [asterisk-users] Channel/Mute
Khaled W. Chehab
- [asterisk-users] AOC (advice of charge) current status.
Roeften
- [asterisk-users] wireless ATA
Jeff LaCoursiere
- [asterisk-users] AGI PHP
James A. Shigley
- [asterisk-users] AMI + AGI for outbound click to dial
J. G.
- [asterisk-users] Asterisk cdr_odbc problems
Atlanticnynex
- [asterisk-users] need BT102 firmware (current version)
Eric Fort
- [asterisk-users] 1.6.1 app_fax: WARNING T.30 ECM carrier not found ??
sean darcy
- [asterisk-users] 64bit: any problems with asterisk?
Andrew Joakimsen
- [asterisk-users] Preferred language for Asterisk AGIs development ?
Kashif Naeem
- [asterisk-users] Dial with MOH
Khaled W. Chehab
- [asterisk-users] "Asterisk cmd MYSQL" app_addon_sql_mysql / performance ?
Julien Chavanton
- [asterisk-users] OT: Polycom handset cord detangler
Dave Fullerton
- [asterisk-users] stop the MOH since asterisk knows that channel is ringing
Khaled W. Chehab
- [asterisk-users] chan_mobile and DTMF
Carlos Ruiz Diaz
- [asterisk-users] John Todd, Moises Silva Speaking At ClueCon 2009
Michael Collins
- [asterisk-users] problems in h323 channels
salzh
- [asterisk-users] Caller information in Web
Kurian Thayil
- [asterisk-users] After transfer context
Administrator TOOTAI
- [asterisk-users] precision of wait dialplan application
Johann Steinwendtner
- [asterisk-users] Questions on X100P/X101P cards
Vincent
- [asterisk-users] ConfBridge versus MeetMe
David Backeberg
- [asterisk-users] Bridge() and Goto() and dialplan contexts, oh my!
David Backeberg
- [asterisk-users] ConfBridge versus MeetMe
Joshua Colp
- [asterisk-users] Cisco 7940 phones become unreachable over VPN after a time
ian at comtek.co.uk
- [asterisk-users] astcc - outgoing call does not hangup properly
Dan Caescu
- [asterisk-users] How to get SIP resposnse codes
Gabriel Ortiz Lour
- [asterisk-users] Where are 2 letter language values defined?
Steve Edwards
- [asterisk-users] Polycom Dialplan Digitmaps
Justin Phelps
- [asterisk-users] Asterisk with Sphinx
Azher Mughal
- [asterisk-users] Cisco 7960G with static config
Azher Mughal
- [asterisk-users] Voice Mail Delete Notification
Brian Alexander
- [asterisk-users] Messaging System
Ricardo Melendez
- [asterisk-users] Sangoma a104d and channel banks
Jim Dickenson
- [asterisk-users] pri errors..
Oguzhan Kayhan
- [asterisk-users] How ro store Reject cause
Venefax
- [asterisk-users] Asterisk sudden crash
Andrew Nowrot
- [asterisk-users] Master.csv
Brent Vrieze
- [asterisk-users] Voicemail Alert
Cary Fitch
- [asterisk-users] Polycom Dialplan Digitmaps
Justin Phelps
- [asterisk-users] How to get meetme participants in dialplan?
Steve Edwards
- [asterisk-users] asterisk-users Digest, Vol 58, Issue 17
Dave Platt
- [asterisk-users] Macro arguments on app_queue
cesar at codinet.com.mx
- [asterisk-users] Voicemail format - no transcode?
Gordon Henderson
- [asterisk-users] QoS & VPN
Brent Davidson
- [asterisk-users] Default dahdi fxs behavior
Jim Dickenson
- [asterisk-users] func_odbc.c: Unable to execute query
arturo arturo
- [asterisk-users] CALL SETUP TIME
research at businesstz.com
- [asterisk-users] Not receiving voicemail message in mailbox
jonas kellens
- [asterisk-users] G.722, 1.4 and IAX trunking ...
Gordon Henderson
- [asterisk-users] "pri show spans" shows nothing
Jim Boykin
- [asterisk-users] Can't GOSUB_RESULT with Dial U() option ...
Olivier
- [asterisk-users] Difference between Transfer and Dial applications
Aurimas Skirgaila
- [asterisk-users] Configuring SIP Trunk
Sathyan M
- [asterisk-users] Numeric Hangup Code
Venefax
- [asterisk-users] DNID Truncated
Jim Boykin
- [asterisk-users] Override sip.conf settings in extensions.conf? Possible?
Josh Fuller
- [asterisk-users] AMOOCON debriefing
randulo
- [asterisk-users] Override sip.conf settings in extensions.conf? Possible?
Josh Fuller
- [asterisk-users] G279 install in 1.6.0.9 ?
Olivier
- [asterisk-users] The efficient way to add MeetMe to pure SIP install ?
Olivier
- [asterisk-users] QoS & VPN
Dave Platt
- [asterisk-users] G279 install in 1.6.0.9 ? [SOLVED]
Olivier
- [asterisk-users] Leg-based CDR proposal updated; Major mods
Steve Murphy
- [asterisk-users] Not receiving voicemail message in mailbox
Dave Walker
- [asterisk-users] Configuring SIP Trunk
Dave Walker
- [asterisk-users] Record all calls
Michelle Dupuis
- [asterisk-users] Storage capacity for call recording
Michelle Dupuis
- [asterisk-users] Possible to add Voice delay?
George Farris
- [asterisk-users] determination of where a call is placed from (physical location)
Eric Fort
- [asterisk-users] VoIP over satellite internet
Eric Fort
- [asterisk-users] IPv6 support?
Andrew Ruthven
- [asterisk-users] Unable to run asterisk CLI commands from php
Sam Hawkin
- [asterisk-users] Professional Setup..
Dave Walker
- [asterisk-users] A side of Digium you may have never seen
randulo
- [asterisk-users] Incompatible changes to asterisk 1.6 MYSQL addon query syntax
John Fawcett
- [asterisk-users] Rusting Snoms?
Christian Stredicke
- [asterisk-users] asterisk blade server
Dean Collins
- [asterisk-users] Special Dialplan
Catalin S.
- [asterisk-users] Professional Setup..
Dave Walker
- [asterisk-users] Building a System.
John F. Ervin
- [asterisk-users] How to write custom functions in AEL2 ,
Olivier
- [asterisk-users] Support of /* */ comments in ael.vim
Olivier
- [asterisk-users] Polycom-330 not displaying line & buddy label?
Yehavi Bourvine
- [asterisk-users] No CDR generated for calls to queues with no agents
Rajkumar S
- [asterisk-users] DTMF received twice
Administrator TOOTAI
- [asterisk-users] Asterisk 1.6.2.0-beta2 Now Available
Asterisk Development Team
- [asterisk-users] VoIP over satellite internet
Josh Fuller
- [asterisk-users] Asterisk w/ Nokia "e" Series Handsets
Cory Andrews
- [asterisk-users] Problems with res_odbc
Daniel - Asterisk
- [asterisk-users] PauseMonitor() Hanging Up Call
Jon Morgan
- [asterisk-users] Ready to put the box on the net
k4rjj at bellsouth.net
- [asterisk-users] Anyone with a working pfSense firewall configuration?
Tim Nelson
- [asterisk-users] [asterisk-user] Which policy for ISDN BRI support in NT/PtMP ?
Kristijan Vrban
- [asterisk-users] Asterisk Manager API Action Originate
Nicholas Blasgen
- [asterisk-users] Help with radius
carl Lougher
- [asterisk-users] Wanting to manipulate SIP response headers
Bruce Ferrell
- [asterisk-users] Is anyone keeping up with the versions?
Thermal Wetland
- [asterisk-users] Hangup()-command does not hang up the line
jonas kellens
- [asterisk-users] enum agi interesting problem
Dan Caescu
- [asterisk-users] no source on cdr logs in some cases!!
Oguzhan Kayhan
- [asterisk-users] Asterisk 1.6 T.38 generation towards a Cisco voice router
Jon Schøpzinsky
- [asterisk-users] Free Fax for asterisk
Markus Weiler
- [asterisk-users] AGI scripts in Groovy, JavaScript, JRuby or PHP running on the Java Virtual Machine
Stefan Reuter
- [asterisk-users] Asterisk doesn't relay remote MOH during hold
Richard Brady
- [asterisk-users] Switchvox
Jeff LaCoursiere
- [asterisk-users] Request for feedback/testing on Multicast RTP Paging
Joshua Colp
- [asterisk-users] Add Monitor application to call suppresses audio
Barry L. Kline
- [asterisk-users] Voicemail and remote directory with SSHFS
Elliot Murdock
- [asterisk-users] Why asterisk changes RTP destination port when it receives first RTP packet in opposite direction despite canreinvite=no
rob.r374 at gmail.com
- [asterisk-users] Asterisk+a2billing for over 10,000 ext
James Mutuku
- [asterisk-users] #-all.gsm
David at ULC
- [asterisk-users] High Volume US Traffic? Claim DIP Compensation!
Marco [voicetermination.org]
- [asterisk-users] Help need to do Lookup from odbc database
carl Lougher
- [asterisk-users] Double dial.
Catalin S.
- [asterisk-users] Problem with Asterisk + TDM410 FXO
Alex Samad
- [asterisk-users] DAHDI [USERUSERINFO]
DHAVAL INDRODIYA
- [asterisk-users] Problem with Asterisk 1.4 and Linksys Spa941/962
Stefan Schmidt
- [asterisk-users] FW: [Dean Collins] Joint BarcampNYC4 sessions?
Dean Collins
- [asterisk-users] SIP error message
Thomas Kenyon
- [asterisk-users] Digium TDM 400 or Openvox A400P
Jonn Taylor
- [asterisk-users] Parked Calls Problem
Brent Vrieze
- [asterisk-users] Goto not matching
michel freiha
- [asterisk-users] Asterisk 1.4.25-rc1 Now Available
Asterisk Development Team
- [asterisk-users] how to avoid call waiting? Or check DIALSTATUS before Dial()?
sean darcy
- [asterisk-users] polycom soundpoint question
Lists
- [asterisk-users] comedian mail
honeyboy at charter.net
- [asterisk-users] Cross-compiling asterisk
Ankit Agarwal
- [asterisk-users] howto build oslec with dahdi-linux-2.1.0.4 or svn?
sean darcy
- [asterisk-users] Help need to do Lookup from odbc database
carl Lougher
- [asterisk-users] how to ignore a ring on a line
Alex Samad
- [asterisk-users] Friday May 15 @12 Noon EDT with Askozia pbx
randulo
- [asterisk-users] Fax t38 capability
Khaled W. Chehab
- [asterisk-users] DTMF Recognition
Timm M.Schneider
- [asterisk-users] DNS host name resolution in iax.conf
Vieri
- [asterisk-users] Zap Transfer
Baskar
- [asterisk-users] Asterisk open source project servers have new names!
Kevin P. Fleming
- [asterisk-users] help a bald guy
Danny Nicholas
- [asterisk-users] Spiral SIP Request problem
amit salunkhe
- [asterisk-users] Strange SIP Activity
M Hulber
- [asterisk-users] meetme dies looking for conf-getconfno
sean darcy
- [asterisk-users] Mediant 1000 audiocodes and Trixbox
Guillermo Garron
- [asterisk-users] What happened here when transfering a call ? Circuit-busy ???
jonas kellens
- [asterisk-users] change AGI script return result
Hristo Benev
- [asterisk-users] change AGI script return result
Hristo Benev
- [asterisk-users] Logging In / Out Agents on Asterisk 6 ???
David Anthony O Reilly
- [asterisk-users] chan_mobile and DTMF
Carlos Ruiz Diaz
- [asterisk-users] Fwd: Asterisk With Cisco Voice Router
Timothy Smith
- [asterisk-users] howto set up persistent dynamic meetme
sean darcy
- [asterisk-users] Agent-Login/out in 1.6
David Anthony O Reilly
- [asterisk-users] Agent-Login/out in 1.6
David Anthony O Reilly
- [asterisk-users] TODAY May 17 Sunday Asterisk VOIP Conference server & Ekiga for BerkeleyTIP
john_re
- [asterisk-users] Correction IRC Channel name - was TODAY May 17 Sunday Asterisk VOIP Conference server & Ekiga for BerkeleyTIP
john_re
- [asterisk-users] Capture "Server" header in SIP reply.
Chris Maciejewski
- [asterisk-users] SHARED() variables and <ZOMBIE> channel
Chris Maciejewski
- [asterisk-users] Can YOU find a trailing parenthesis?
sean darcy
- [asterisk-users] Calls Declined
David at ULC
- [asterisk-users] Switchvox
Dave Walker
- [asterisk-users] Manager API in PHP
Olivier
- [asterisk-users] Open source SIP client
DHAVAL INDRODIYA
- [asterisk-users] Problem with Free Fax For Asterisk
Trevor Hammonds
- [asterisk-users] Panasonic SIP Phone
Si Tai Fan
- [asterisk-users] no video by Originate
salzh
- [asterisk-users] ${HANGUPCAUSE} is not printed when call ends or is interrupted
jonas kellens
- [asterisk-users] Some direction for creating sound files for Asterisk
Jonathan Moore
- [asterisk-users] meetme
Jeff LaCoursiere
- [asterisk-users] Number of max SIP calls.
Ankit Agarwal
- [asterisk-users] callcenter / dialer / predictive dialer / vicidial program is now open
ContactTel Business
- [asterisk-users] From 1.4 to 1.6.0
Miguel Molina
- [asterisk-users] 1.6.1.0 with Realtime Mysql
Carlos Chavez
- [asterisk-users] Realtime Static on 1.6.1.0
Carlos Chavez
- [asterisk-users] Playing audio messages to the callee.
lucky goyal
- [asterisk-users] OT: SIP hardphone with multi-color BLF
Olivier
- [asterisk-users] SPA941
Dimitris Counalakis
- [asterisk-users] How to access voicemail from deskphone
amit salunkhe
- [asterisk-users] Rusting Snoms?
Christian Stredicke
- [asterisk-users] Unable to make outbound calls
Kal Feher
- [asterisk-users] Dialplan Priorities and Sort Order...
Tim Nelson
- [asterisk-users] Question
Venefax
- [asterisk-users] Ghost ??
David at ULC
- [asterisk-users] Alternative to Adobe Audition 3 for G723 > G711 uLaw ? (old Cool Edit Pro)
Jason Aarons (US)
- [asterisk-users] cdr record disposition always FAILED
John Regal
- [asterisk-users] Feature request: "database show" from manager API
Olivier
- [asterisk-users] Dialplan matching problem
michel freiha
- [asterisk-users] announcement: chan_nms - channel driver for NMS Communications hardware
Arkadi Shishlov
- [asterisk-users] Hang at 5:34 pm EST
David at ULC
- [asterisk-users] What codec/sample rate/resolution...?
Jason Aarons (US)
- [asterisk-users] Manager ExtensionState function
Azher Mughal
- [asterisk-users] Asterisk CCM, CME Integration
Arun Kumar
- [asterisk-users] FaxIn problems
srinivas Antarvedi
- [asterisk-users] TC400
Kashif Ali
- [asterisk-users] Feature request: "database show" from manager API [SOLVED]
Olivier
- [asterisk-users] Problems receiving some faxes in T.38
Santiago Gimeno
- [asterisk-users] FritzBox 7270
Manoj Panicker - FOES
- [asterisk-users] Pickup with *8 is not working...
jonas kellens
- [asterisk-users] Queue and Dial operation - Common Variables?
Kurian Thayil
- [asterisk-users] dtmf=info and canreinvite=yes
Philipp Kempgen
- [asterisk-users] asterisk crash on DAHDI error: No more room in scheduler
Hose
- [asterisk-users] Step-by-Step Asterisk and MeetMe Help
Josh Fuller
- [asterisk-users] How to detect switch to voicemail when calling to mobile phone
Michel Verbraak
- [asterisk-users] asterisk memory (issue)
Deepak
- [asterisk-users] MeetMe - Different pin for different user
Jim Boykin
- [asterisk-users] play with varibles
BERGANZ François
- [asterisk-users] Macro with DIALSTATUS
Azher Mughal
- [asterisk-users] Do I need a SIP Proxy for this?
Jonathan Moore
- [asterisk-users] DAHDI fun and games
Danny Nicholas
- [asterisk-users] ...is circuit busy message
John Regal
- [asterisk-users] 1.4.24.1 -> 1.6.0.9: segfault
sean darcy
- [asterisk-users] Voicemail playback NEWEST first vs. OLDEST first
Karl Fife
- [asterisk-users] Bridging INBOUND PRI to OUTBOUND PRI fails with Monitor()
Barry L. Kline
- [asterisk-users] PSTN Connection
Manoj Panicker - FOES
- [asterisk-users] interruption in queue
Rilawich Ango
- [asterisk-users] Polycom Productivity Suite
Matt Darnell
- [asterisk-users] MeetMe not working with GSM codec?
Chris Maciejewski
- [asterisk-users] Jitter buffer question
Ondrej Valousek
- [asterisk-users] playing media(moh,prompts) from flash player
marek cervenka
- [asterisk-users] reg static build
sasirekha jaganathan
- [asterisk-users] Jitter buffer question
Ondrej Valousek
- [asterisk-users] Zaptel Error
Farooq Hussain
- [asterisk-users] Calling party category
equis software
- [asterisk-users] Page/Intercom problem
Brent Vrieze
- [asterisk-users] Asterisk 1.4.25 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk-Addons 1.6.0.2 Now Available
Asterisk Development Team
- [asterisk-users] calls stuck in AMD even after analysis time
Roi Stork
- [asterisk-users] Cheapest price to cuba route !!!
ContactTel Business
- [asterisk-users] Free Fax for Asterisk Receiving problem
Danny Nicholas
- [asterisk-users] "...is circuit-busy" message
Dave Walker
- [asterisk-users] Parsing Asterisk's .conf files from Perl, Java or PHP file
Olivier
- [asterisk-users] ...is circuit busy message
Dave Walker
- [asterisk-users] Error ON SIP Incoming TOS
DHAVAL INDRODIYA
- [asterisk-users] Can't get G.726 to work.
Chris Maciejewski
- [asterisk-users] rasterisk r processes take the rest of my cpu
Giorgio Incantalupo
- [asterisk-users] /etc/asterisk/startup.d
Philipp Kempgen
- [asterisk-users] VOICEMAIL : I've tried a lot but mailing through Asterisk is just not working...
jonas kellens
- [asterisk-users] Indications.conf and tone generation volume
Lee Spenadel
- [asterisk-users] Alison Keenan (free British English voice)
Mark Phillips
- [asterisk-users] No response to our critical packet problem
James Lamanna
- [asterisk-users] DTMF
David at ULC
- [asterisk-users] How to stop a background music
Noel R. Morais
- [asterisk-users] visp multiaccount + firewall configuration problem
Alex Samad
- [asterisk-users] Faxing issues
Todd S
- [asterisk-users] integrating CTI
peace keeper
- [asterisk-users] Asterisk automatically closing the file descriptor
arnuld uttre
- [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
Dunc
- [asterisk-users] 1.6.0.9 sip.c: "Serious Network Trouble" ??
sean darcy
- [asterisk-users] 1.6.0.9: Unknown signalling method 'pri_cpe' ??
sean darcy
- [asterisk-users] [OT]I like this community
Rony Ron
- [asterisk-users] RPID on SNOM phones?
Yehavi Bourvine
- [asterisk-users] Duplicate DTMF digits
AC
- [asterisk-users] Can I run two instances of asterisk
Julian Lyndon-Smith
- [asterisk-users] Basic Config
Farooq Hussain
- [asterisk-users] Connected Number on incoming calls with mISDN
Andreas Ruf
- [asterisk-users] Problem running Dahdi
Mike
- [asterisk-users] SIP Trunk groups
Mariano Lecuona
- [asterisk-users] New tutorial: storing audio recordings per day
Lenz Emilitri
- [asterisk-users] DNS issues again
Joseph L. Casale
- [asterisk-users] howto store local exchange prefixes ?
sean darcy
- [asterisk-users] Placing a MWI on-off call...
eric weaver
- [asterisk-users] asterisk-addon 1.6.1 problem
Rilawich Ango
- [asterisk-users] h extension and channel variables
Thomas Kenyon
- [asterisk-users] Logging calls made/lost
Andreas-Johann Ulvestad
- [asterisk-users] Bandwidth management and ADSL router
bilal ghayyad
- [asterisk-users] Converting Cisco 7961 to SIP
Darrin Henshaw
- [asterisk-users] Hanging up a call by DTMF
abdelkader
- [asterisk-users] A problem in playing sound files
abdelkader
- [asterisk-users] Maximum cable length for analog phone from FXS port
asterisk-users at rogg.is
- [asterisk-users] FXS
Diogo Saad
- [asterisk-users] CDR after SIP blind transfer.
Chris Maciejewski
- [asterisk-users] SIP over VPN
Marco Sambo
- [asterisk-users] How to register with TCP transport ?
Olivier
- [asterisk-users] Suggest good calling service for London
Kashif Naeem
- [asterisk-users] Bandwidth management and ADSL router
bilal ghayyad
- [asterisk-users] No Voice - only "noisy audio"
Diogo Saad
- [asterisk-users] STUN setting in Asterisk 1.6.X
Carlos Chavez
- [asterisk-users] Strange message in CLI
Joseph L. Casale
- [asterisk-users] Bug or feature in 1.6.1 (Was: How to register with TCP transport) ?
Olivier
- [asterisk-users] Indications.conf and tone generation volume
Lee Spenadel
- [asterisk-users] Fax Machines across carrier SIP trunk? General recommendation?
Jason Aarons (US)
- [asterisk-users] multiple bind ports with TCP and UDP
Olivier
- [asterisk-users] Bandwidth management and ADSL Router
bilal ghayyad
- [asterisk-users] Call in progress tones
Mikel Lindsaar
- [asterisk-users] PHP AGI Problems
Atlanticnynex
- [asterisk-users] Asterisk memory problems
Ikin Wirawan
- [asterisk-users] Delay and Zombie Channels Problem
Niko P Kusumah
- [asterisk-users] DAHDI and hangup issue when playing the IVR
Tharanga
- [asterisk-users] AstDB wildcards
Geoff Lane
- [asterisk-users] stucked calls in asterisk 1.4
Stefan Schmidt
- [asterisk-users] Pressing number 2 in dialplan
Elliot Murdock
- [asterisk-users] No full duplex communication ?
jonas kellens
- [asterisk-users] TDM400P in PCI-X Slot
Michael C. Cambria
- [asterisk-users] 1.6.0.9: Now "Unable to create ... 'DAHDI'"
sean darcy
- [asterisk-users] SIP Trunk groups
Mariano Lecuona
- [asterisk-users] setting CDR values on failed calls
John Regal
- [asterisk-users] Playtones Volume
Lee Spenadel
- [asterisk-users] Auto-congesting call due to slow response
Alexander Topolanek
- [asterisk-users] Domains
Dave Walker
- [asterisk-users] Reg AsteriskNow 1.5 Beta Release
sasirekha jaganathan
- [asterisk-users] asterisk 1.4.X, T.38 and NAT
Antoine Megalla
- [asterisk-users] probably an rtfm but... need to dial out to 2 PSTN lines from AMI
John Millican
- [asterisk-users] SIP CALL ENCRYPTION
research at businesstz.com
- [asterisk-users] SIP CALL ENCRYPTION
research at businesstz.com
- [asterisk-users] Friday at 12 Noon EDT: Jim Van Meggelen on the VoIP Users Conference
randulo
- [asterisk-users] zaptel installation
Jerry Geis
- [asterisk-users] Best Current Release for Long Term Use
Dave Walker
- [asterisk-users] Call telco transfer q931
Andres Gomez
- [asterisk-users] To: Field
Charles Solar
- [asterisk-users] CAll-limit or incominglimit ?????
Yuri
- [asterisk-users] asterisk 1.6.1.0 and dial plan changes
Tharanga
- [asterisk-users] how to detect dtmf in meetme
robert
- [asterisk-users] how to detect dtmf in meetme
robert
- [asterisk-users] connection fail between Service provider's proxy server and my asterisk server
김무성
- [asterisk-users] AMI and Originate on 1.6.0.5
DHAVAL INDRODIYA
- [asterisk-users] IAX2 trunking with Older Asterisk version ?
Tharanga
- [asterisk-users] regarding to field of accountcode
Rilawich Ango
- [asterisk-users] Logging into queue homed off remote system
Ekelund, Bryan
- [asterisk-users] How to read values from another channel ?
Olivier
- [asterisk-users] Attended transfer and dialplan
Olivier
- [asterisk-users] dial out context from incoming iax trunk
Oguzhan Kayhan
- [asterisk-users] asterisk crash on DAHDI error: No more room in scheduler
Allan Oepping
- [asterisk-users] SIP CALL: RTP ENCRYPTION
research at businesstz.com
- [asterisk-users] Queue - Multiple Transfer
Kurian Thayil
- [asterisk-users] compile error for chan_h323
salzh
- [asterisk-users] Understanding Call Handling In Asterisk
varun.rapelly at spectross.com
- [asterisk-users] SIP CALL: RTP ENCRYPTION
research at businesstz.com
- [asterisk-users] Simplex voice on TDM410P
Nathanial A. Byrnes
- [asterisk-users] Problem T.38
Daviramos Roussenq Fortunato
- [asterisk-users] POS Modems
adriano.furtado at terra.com.br
- [asterisk-users] question about reinvite
Alex Samad
- [asterisk-users] Multile IP addresses for SIP device
Elliot Murdock
- [asterisk-users] h323 guide for asterisk
Tamer Higazi
- [asterisk-users] An outside Caller ID not shown,
peace keeper
- [asterisk-users] Network settings and quality of voice
bilal ghayyad
- [asterisk-users] Ekiga, Twinkle and from where to start with open source
bilal ghayyad
- [asterisk-users] Asterisk 1.4.25 and zapata.conf
bilal ghayyad
- [asterisk-users] Suddenly the voice became garbage (like robot) using Asterisk 1.4.19.2
bilal ghayyad
Last message date:
Sun May 31 23:43:32 CDT 2009
Archived on: Wed Jun 3 21:44:28 CDT 2009
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