January 2011 Archives by thread
Starting: Fri Dec 31 22:34:12 CST 2010
Ending: Mon Jan 31 21:09:05 CST 2011
Messages: 1108
- [asterisk-users] Base memory usage
Jeremy Kister
- [asterisk-users] Base memory usage
Gilles
- [asterisk-users] Log and forward calls to cellphone?
Gilles
- [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)
bilal ghayyad
- [asterisk-users] Cisco IP Phones and AVAYA IP Phones: How to configure in Asterisk
bilal ghayyad
- [asterisk-users] DIALSTATUS on CANCEL
Bryant Zimmerman
- [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)
Bryant Zimmerman
- [asterisk-users] CDR Questions
Mike Diehl
- [asterisk-users] Forward voicemail not working
duane.larson at gmail.com
- [asterisk-users] digim tdm2400p fxo fake answer supervision problem.
Muhammad Usman
- [asterisk-users] changed datadir
Nicholas Hart
- [asterisk-users] VoIP PoE phones for restaurant
Andy Graybeal
- [asterisk-users] Queue announce parameter problem
Eduardo Lobo Blanco
- [asterisk-users] Clarification on DAHDI Fax Detection
Tom Rymes
- [asterisk-users] 1.8 MIBs
Kirill Katsnelson
- [asterisk-users] Voicemail Forwarding
--[ UxBoD ]--
- [asterisk-users] Go from CALLINGout to just CALLING
Jonas Kellens
- [asterisk-users] OT - Contact center - How to Gmail-like label to incoming email
Olivier
- [asterisk-users] Queues, priorities and (miscalculated) holdtimes
Daniel Tryba
- [asterisk-users] MOH problems (asterisk 1.4.38)
Earl Terwilliger
- [asterisk-users] Fwd: Announce: telepathy-ring 2.1.1
Steve Totaro
- [asterisk-users] DAHDI and dialdebounce
Tom Rymes
- [asterisk-users] problems inserting dahdi modules using Debian Leni
covici at ccs.covici.com
- [asterisk-users] VoIP PoE phones for restaurant (kitchen)
Jeff LaCoursiere
- [asterisk-users] VoIP PoE phones for restaurant (kitchen)
mgraves at mstvp.com
- [asterisk-users] Asterisk Outlook integration
Bruce B
- [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
Bruce B
- [asterisk-users] Add Privacy: id to SIP-invite
Jonas Kellens
- [asterisk-users] Add Privacy: id to SIP-invite
Bryant Zimmerman
- [asterisk-users] Blind Transfer not working - 1.4.38
Ishfaq Malik
- [asterisk-users] DTMF-troubles with Snom
Jonas Kellens
- [asterisk-users] TE420 issue: card 0 span N: isr2=XX isr3=Y
Tony Mountifield
- [asterisk-users] problems inserting dahdi modules using Debian Leni
Dave Platt
- [asterisk-users] Asterisk replying to wrong port for NOTIFY messages
James Lamanna
- [asterisk-users] TDM410 and DSL
Cassius Smith
- [asterisk-users] using google for vm transcripts
sean darcy
- [asterisk-users] dtmf-troubles with Snom
Jonas Kellens
- [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
mgraves at mstvp.com
- [asterisk-users] SILK codec
Edwin Lam
- [asterisk-users] system lockup when going into conference
covici at ccs.covici.com
- [asterisk-users] Anyone have Festival application working?
Ian Pilcher
- [asterisk-users] Force different codecs on call base
Daniel Tryba
- [asterisk-users] Call queues on load-balanced asterisks
Pan B. Christensen
- [asterisk-users] Channel name changed in asterisk 1.8
Arjan Kroon | Mobillion
- [asterisk-users] DTMF-troubles with Snom
Bryant Zimmerman
- [asterisk-users] AGI->Macro w/Agruments
William Stillwell
- [asterisk-users] AstLinux 0.7.5 released
Darrick Hartman (lists)
- [asterisk-users] Grandstream GXE2504A codec disable option
amit salunkhe
- [asterisk-users] Mail list Woes?
jonp at inline.net
- [asterisk-users] Mail list Woes?
William Stillwell
- [asterisk-users] Call parking question
Chris Gentle
- [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
mgraves at mstvp.com
- [asterisk-users] How to check a number online or offline
Phuong Hoang
- [asterisk-users] Call Back on Busy
Ron
- [asterisk-users] New Dahdi error
cjwstudios
- [asterisk-users] slow response to INVITE
Ron
- [asterisk-users] Fix Fake Answer Supervision In asterisk1.6
Muhammad Usman
- [asterisk-users] Do I need a sip proxy?
Bruce B
- [asterisk-users] asterisk fax problem
Oguzhan Kayhan
- [asterisk-users] How to check a number online or offline
Phuong Hoang
- [asterisk-users] Call queues on load-balanced asterisks
Pan B. Christensen
- [asterisk-users] OpenVPN + SIP configuration?
Gilles
- [asterisk-users] Show voicemail in GUI
Jonas Kellens
- [asterisk-users] Call queues on load-balanced asterisks
Thomas Liu
- [asterisk-users] Using the Telco Call Transfer Features.
Jeff B
- [asterisk-users] Fail2Ban & CSF
Jonas Kellens
- [asterisk-users] Problems with ZAP Channels
Antonio Modesto
- [asterisk-users] Queue periodic announce...
Carlos Chavez
- [asterisk-users] SetVar Warning
Gary Kuznitz
- [asterisk-users] Call hung up?
Gary Kuznitz
- [asterisk-users] Polycom Blf / Directed Pickup
Mark Murawski
- [asterisk-users] Fax stopped working when upgrading to 1.8.2
magnus.b at inputinterior.se
- [asterisk-users] queue_log in MySQL database
Jonas Kellens
- [asterisk-users] CallerID and URL pop up for windows...
Carlos Chavez
- [asterisk-users] 5-7 second delay in connecting outgoing FXO calls
ftarz
- [asterisk-users] Asterisk+h324m gateway issue
pankaj pandey
- [asterisk-users] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'
Jonas Kellens
- [asterisk-users] Selecing the E1 cards for the call center
bilal ghayyad
- [asterisk-users] Ghost ringing
jfratantoni at iswan.net
- [asterisk-users] Why are 4 ports used for a single call?
Bruce B
- [asterisk-users] Asterisk 1.4.39 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk 1.6.2.16 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk 1.8.3 Now Available
Asterisk Development Team
- [asterisk-users] Spectralink 8002
Jonathan C. Bailey
- [asterisk-users] Why are 4 ports used for a single call?
Bruce B
- [asterisk-users] Tools to Monitor Asterisk Servers and VMs
Bruce B
- [asterisk-users] Top Posting
Don Kelly
- [asterisk-users] Bruce B
Tim Nelson
- [asterisk-users] Asterisk stops responding
Carlos Chavez
- [asterisk-users] Problem with chan_dahdi and conferencing
covici at ccs.covici.com
- [asterisk-users] Music on Hold not working?
Gary Allen
- [asterisk-users] Sound quality issue
Cédric Lemarchand
- [asterisk-users] T.38 Digium Fax Driver Success on Fail
Elliot Murdock
- [asterisk-users] Selecting the E1 cards for the call
bilal ghayyad
- [asterisk-users] res_fax_digium.so crashing
Jeremy Kister
- [asterisk-users] app_calendar and SSL
--[ UxBoD ]--
- [asterisk-users] 'Bad authorization' error with Asterisk 1.8
Arie Goldfeld
- [asterisk-users] Occasional robotic sound while call in progress
Michelle Dupuis
- [asterisk-users] Max call duration
Michelle Dupuis
- [asterisk-users] Top Posting
Andrew Thomas
- [asterisk-users] Sound quality issue
Andrew Thomas
- [asterisk-users] Can I know if a call is transffered to asterisk
ishagh ouldbah
- [asterisk-users] Top Posting
Andrew Thomas
- [asterisk-users] Top Posting
Andrew Thomas
- [asterisk-users] Top Posting
Andrew Thomas
- [asterisk-users] Asterisk Security Releases: AST-2011-001
Asterisk Development Team
- [asterisk-users] SIP Originate on 1.8.1.1
Carlos Chavez
- [asterisk-users] Calling rules
Vitor Carlos Flausino
- [asterisk-users] 1.8.2: dahdi-2.4: calls dropping
sean darcy
- [asterisk-users] chan_sip.c: Failed to parse contact info
Nick Ustinov
- [asterisk-users] Asterisk extension not found problem...
abhinav anand
- [asterisk-users] Make ConfBridge hang up on last participant
Ian Pilcher
- [asterisk-users] No RTP Engine problem in 1.8.2
Paradise Dove
- [asterisk-users] How to detect line tone?
Massimo Nuvoli
- [asterisk-users] sip dos question
adamk at 3a.hu
- [asterisk-users] Asterisk 1.8.2 and digium yum repositories
Ishfaq Malik
- [asterisk-users] Asterisk fail over. From IP rewrite issues
Peter den Hartog
- [asterisk-users] audiohook.c: Write factory 0x153cf678 was pretty quick last time, waiting for them
Jonas Kellens
- [asterisk-users] agi dial termination cause ?
mancyborg at gmail.com
- [asterisk-users] IAX between 1.6 and 1.8 has bad voice quality
Carlos Chavez
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] Cross Queue Priorities
Nick Brown
- [asterisk-users] Internode weirdness
Da Rock
- [asterisk-users] Using asterisk and icecast for live audio streaming.
Goke M Aruna
- [asterisk-users] context problem
Jonas Kellens
- [asterisk-users] OT - TTS in spanish
Olivier
- [asterisk-users] Accessing a 'user' variable via. dialplan.
Andrew Thomas
- [asterisk-users] ReceiveFax
Bryant Zimmerman
- [asterisk-users] Accessing a 'user' variable via. dialplan.
Andrew Thomas
- [asterisk-users] context problem
Andrew Thomas
- [asterisk-users] Mailing list question
Andrew Thomas
- [asterisk-users] Mailing list question
Andrew Thomas
- [asterisk-users] Mailing list question
Andrew Thomas
- [asterisk-users] Mailing list question 2
Andrew Thomas
- [asterisk-users] Mailing list question
Andrew Thomas
- [asterisk-users] Mailing list question 2
Andrew Thomas
- [asterisk-users] Polycom 500 / MWI
Brian C. Huffman
- [asterisk-users] context problem
Dave Platt
- [asterisk-users] Asterisk to asterisk t.38
Amit Nepal
- [asterisk-users] Asterisk 1.8.2.2 Now Available (Security Release)
Asterisk Development Team
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] Asterisk to asterisk t.38
Bryant Zimmerman
- [asterisk-users] SIP client floods port 5060 and gets blocked
Julian Yap
- [asterisk-users] Polycom SoundPoint IP 650 freezes on boot after adding just one custom ringtone
Marco Lechner - FOSSGIS e.V.
- [asterisk-users] Unable to receive calls (inbound)
Vitor Carlos Flausino
- [asterisk-users] Mailing list question
Andrew Thomas
- [asterisk-users] Where are stored the CDR's?
Vitor Carlos Flausino
- [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?
Zeeshan Zakaria
- [asterisk-users] Inbound routes
Vitor Carlos Flausino
- [asterisk-users] MOH and parking
Andrew Thomas
- [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?
Zeeshan Zakaria
- [asterisk-users] Force Dahdi modules to load
Pablo Schuhwerk
- [asterisk-users] Queues with ringinuse=yes
Vinícius Fontes
- [asterisk-users] Channel in an unkown state
Vitor Carlos Flausino
- [asterisk-users] waitforsilence changed after upgrade to 1.6
Mike Diehl
- [asterisk-users] spandsp download
Bryant Zimmerman
- [asterisk-users] Asterisk stops responding
Carlos Chavez
- [asterisk-users] end a call after a specific time period
ABBAS SHAKEEL
- [asterisk-users] Dialplan to bridge 2 legs?
Michelle Dupuis
- [asterisk-users] Info on using LDAP with Asterisk?
Jeff B
- [asterisk-users] B410P: DAHDI BRI PTMP HDLC Abort (6) on Primary D-channel of span 1 (TEI Errors)
Matt Riddell
- [asterisk-users] Outgoing FXO calls have no audio with callprogress=no
Frank Tarczynski
- [asterisk-users] Asterisk on Debian Lenny with timerfd
RR
- [asterisk-users] Crossover cable for E1 ?
A J Stiles
- [asterisk-users] Unable to insert cdr-data into mysql-DB
Jonas Kellens
- [asterisk-users] Asterisk on Debian Lenny with timerfd
Dave Platt
- [asterisk-users] ReceiveFAX issue.
Bryant Zimmerman
- [asterisk-users] U-verse DTMF tuning for Zaptel
Steve Edwards
- [asterisk-users] extconfig, realtime, and SIP
Richard Kenner
- [asterisk-users] Unknow "T" callerid
Jose Flores Galicia
- [asterisk-users] Asterisk on Debian Lenny with timerfd
Dave Platt
- [asterisk-users] Unable to insert cdr-data into mysql-DB
Andrew Thomas
- [asterisk-users] MOH and parking
Andrew Thomas
- [asterisk-users] SIP RTP streams
Da Rock
- [asterisk-users] ReceiveFAX issue.
Bryant Zimmerman
- [asterisk-users] Problem registering two (and more) sip trunks
Luis Silva
- [asterisk-users] regarding quit, exit and stop now in asterisk
viswavardhanreddy karna
- [asterisk-users] Help determining SpanDSP version
Tom Rymes
- [asterisk-users] ReceiveFAX issue.
Bryant Zimmerman
- [asterisk-users] Recommended Windows client to display CID?
Gilles
- [asterisk-users] Caching CALLERID(dnid)
Arjan Kroon | Mobillion
- [asterisk-users] Pickup local/.... not working
Jonas Kellens
- [asterisk-users] ReceiveFAX issue.
Bryant Zimmerman
- [asterisk-users] Regarding error in Asterisk dail plan:
Bryant Zimmerman
- [asterisk-users] Really wacky problem with internal extensions.
Ernie Dunbar
- [asterisk-users] Recommended Windows client to display CID?
Asterisk
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] Asterisk 1.8.2.3 Now Available
Asterisk Development Team
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] Outgoing FXO calls have no audio with callprogress=no
Frank Tarczynski
- [asterisk-users] Asterisk 1.8 and Cisco 7920
MrHanMan
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] Callback when available
Harel Cohen
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] Anybody ever see this before?
William Stillwell
- [asterisk-users] Multi-Tenant
Amardeep Rana
- [asterisk-users] OT: VoIP Users Conf Feb 4 with LifeSize
Michael Graves
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] A1200P comments?
Mike Diehl
- [asterisk-users] chan_sip bug? (Asterisk 1.4)
Jian Gao
- [asterisk-users] RTP keepalive doesn't work
Ryan Tucker
- [asterisk-users] SendFAX dialplan example
magnus.b at inputinterior.se
- [asterisk-users] How to update sound files?
Сикорский Сергей
- [asterisk-users] CDR issue - Problem logging CDR(userfield) in Master.csv
Athanasia Tsertou
- [asterisk-users] A1200P comments?
Valer Nur
- [asterisk-users] How to disable srtp in asterisk 1.8.2.3?
Miguel Baptista
- [asterisk-users] Asterisk 1.8.2 - TLS, user certificate
Gilles 李乐
- [asterisk-users] Asterisk Scenary
José Luis Hernández Ramos
- [asterisk-users] Reducing number of Asterisk processes?
Gilles
- [asterisk-users] Determine When Call Is Picked Up In Queue
Joseph Begumisa
- [asterisk-users] faxter
Pezhman Lali
- [asterisk-users] Losing registration - ast 1.4.39 and innomedia 6328-2Re
Brian C. Huffman
- [asterisk-users] Error compiling Dahdi: invalid use of undefined type struct module
David Cunningham
- [asterisk-users] Issue with Asterisk not hanging up second leg when first leg hangs up
Dovid Bender
- [asterisk-users] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
Benny Amorsen
- [asterisk-users] save the calls with asterisk
salaheddine elharit
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] Calling Directory app from AGI
Mike Diehl
- [asterisk-users] Newbie Question...
Piotr Górski
- [asterisk-users] res_fax
Bryant Zimmerman
- [asterisk-users] [OT] Streaming video on variable bandwidth connection?
Tim Dobson
- [asterisk-users] regarding error in asterisk
viswavardhanreddy karna
Last message date:
Mon Jan 31 21:09:05 CST 2011
Archived on: Mon Apr 11 11:29:20 CDT 2011
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