September 2010 Archives by thread
Starting: Wed Sep 1 03:23:50 CDT 2010
Ending: Thu Sep 30 15:44:31 CDT 2010
Messages: 1240
- [asterisk-users] Asterisk routing to SoftSwitch
Pratik Shrestha
- [asterisk-users] Logging the CID from the Privacy Manager
Jaap Winius
- [asterisk-users] Freepbx + Asterisk problem - NEED HELP
Mehmet Kuzulugil
- [asterisk-users] 3Com 3102 Phones
Barry Fawthrop
- [asterisk-users] ChanSpy getting piled up
Rushikesh
- [asterisk-users] * and mj
Jeff Jones
- [asterisk-users] * and mj
Infra
- [asterisk-users] MOH in the middle of the call
Dario Quiroz
- [asterisk-users] help with dialplan
Steve Murphy
- [asterisk-users] libpri 1.4.11.4 Now Available
Asterisk Development Team
- [asterisk-users] NCS - Cablemodem
Protectix - IT Solutions
- [asterisk-users] asterisk-1.8 problem with one-way audio with no nat
covici at ccs.covici.com
- [asterisk-users] IAX2 calls getting rejected without a CAUSE CODE. How to debug this?
bruce bruce
- [asterisk-users] Dahdi issue on sangoma A200
asteriskguru asteriskguru
- [asterisk-users] Call Recording Questions
Dan Journo
- [asterisk-users] Voicemail - disable * 0 and #
Paddy Grice
- [asterisk-users] Google Voice-like feature.
Ken D'Ambrosio
- [asterisk-users] Google Voice-like feature.
Ken D'Ambrosio
- [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost?
bruce bruce
- [asterisk-users] How to finish an AGI
Danny Dias
- [asterisk-users] How to create a coredump for Asterisk
Thorolf Godawa
- [asterisk-users] asterisk 1.6.2.11 freezes the server
George
- [asterisk-users] Channel Signalling
Arnaldo Giacomitti Junior
- [asterisk-users] Asterisk processing URI's
Sascha Ferley
- [asterisk-users] Asterisk failing when recording calls
Carlos Chavez
- [asterisk-users] not succeeding to hide callerid with outbound calls
Joost Kuif | Mobillion
- [asterisk-users] Wanted: UK-specific hardware recommendations (FXO and FXS)
Roger Burton West
- [asterisk-users] openvz
mattias
- [asterisk-users] How to use MYSQL(Set timeout x)
F B
- [asterisk-users] [draft] DAHDI-linux & DAHDI-tools 2.4.0 Release Announcement
Asterisk Development Team
- [asterisk-users] Faxes
dave george
- [asterisk-users] How to tell if there is a transfer from CDR?
Carlos Chavez
- [asterisk-users] openvz
Zeeshan Zakaria
- [asterisk-users] Wanted: UK-specific hardware recommendations (FXOand FXS)
--[ UxBoD ]--
- [asterisk-users] Manuplating Queue
Timothy Smith
- [asterisk-users] fast busy out?
Thomas Perron
- [asterisk-users] Vitelity offline?
Roger Marquis
- [asterisk-users] Possible malformed G729B - SID (VAD/DTX) frames from carrier endpoint ?
Vikram Ragukumar
- [asterisk-users] How to tell if there is a transfer from CDR?
Bryant Zimmerman
- [asterisk-users] How to tell if there is a transfer from CDR?
Bryant Zimmerman
- [asterisk-users] Registering and initiating a SIP call without a SIP client
Gautam Desai
- [asterisk-users] evil disconnect of call with cisco 1760
Jeremy Kister
- [asterisk-users] Is it possible to keep both call legs live after Dial()
Barry O'Donovan
- [asterisk-users] Macro when calling cellphone (GSM) + silence when connecting
Jonas Kellens
- [asterisk-users] Dial timeout and SIP 302 Moved Temporarily
Olivier
- [asterisk-users] Going to go out on a limb here - regarding Vonage
GlenM
- [asterisk-users] Asterisk stops processing calls...
Carlos Chavez
- [asterisk-users] How are shared variables destroyed ?
Olivier
- [asterisk-users] What can make G.729a codec hostid change?
Barry Miller
- [asterisk-users] not succeeding to hide callerid with outbound calls
Joost Kuif | Mobillion
- [asterisk-users] voice mail system
Frenette, Rob
- [asterisk-users] Macro when calling cellphone (GSM) + silence when connecting
Jonas Kellens
- [asterisk-users] What can make G.729a codec hostid change?
Dave Platt
- [asterisk-users] 5-7 second connection delay in outgoing FXO calls
Frank Tarczynski
- [asterisk-users] Call Center: scripting for call routing, reporting, login and logout, CTI
bilal ghayyad
- [asterisk-users] Losing first DTMF digit (with ASR)
Richard Kenner
- [asterisk-users] Losing first DTMF digit (with ASR)
Bryant Zimmerman
- [asterisk-users] Solving the CDR mess of attended transfers
Fabiano Carlos Heringer
- [asterisk-users] Losing first DTMF digit (with ASR)
Bryant Zimmerman
- [asterisk-users] 5-7 second connection delay in outgoing FXO calls
Frank Tarczynski
- [asterisk-users] rtcp to cdr for calls from dahdi to sip
Dmitry Melekhov
- [asterisk-users] Requirement or just Best Practice
Danny Nicholas
- [asterisk-users] asterisk 1.8 Calendar
Adrià Vidal
- [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtime mysql
Jonas Kellens
- [asterisk-users] Queue/Dial Recording - Capture answering channel name.
Steve Davies
- [asterisk-users] Problem with new AEX800 card dying because of interrupt problems
Christian Weeks
- [asterisk-users] IPSec on asterisk
Deepika Nijhawan
- [asterisk-users] Asterisk 1.8.0-beta5 Now Available
Asterisk Development Team
- [asterisk-users] IPSec on asterisk
Paul Belanger
- [asterisk-users] Thailand DID
Eric Smith
- [asterisk-users] Asterisk 1.6 and fax
Stanislav Korsei
- [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again..
dashy dude
- [asterisk-users] IPSec on asterisk
Deepika Nijhawan
- [asterisk-users] syntax error, unexpected '<token>'
Jonas Kellens
- [asterisk-users] IPSec on asterisk
Paul Belanger
- [asterisk-users] Mirroring or other arangement to secure *
hbk
- [asterisk-users] Mirroring or other arangement to secure *
Matthew J. Roth
- [asterisk-users] Set channel variable from within other channel
Jonas Kellens
- [asterisk-users] info about application not available asterisk 1.6.2.11
Jonas Kellens
- [asterisk-users] vegastream 50 BRI-s latest firmware ?
mancyborg at gmail.com
- [asterisk-users] Issues with in-call DTMF using Broadvox and Level 3
Bryant Zimmerman
- [asterisk-users] VoIP friendly Internet providers in Dallas and Philadelphia
bruce bruce
- [asterisk-users] Archive of security advisories?
Carlos Chavez
- [asterisk-users] Curious what 'early media' is in terms of Answer()
Hose
- [asterisk-users] DAHDI fxstest?
Tim Nelson
- [asterisk-users] 1.6.2.11 realtime sip registrations disappear from DB
Jonas Kellens
- [asterisk-users] problem with iax call (chan unavailable)
iscario at free.fr
- [asterisk-users] Anyone can share their experience about Thomson TG784 wireless router/ATA?
bruce bruce
- [asterisk-users] Cisco or Linksys WRP400 reliability?
bruce bruce
- [asterisk-users] A way to check against a list of numbers?
Hose
- [asterisk-users] SIP softphones answer but do not connect...
Carlos Chavez
- [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
bruce bruce
- [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Zeeshan Zakaria
- [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Zeeshan Zakaria
- [asterisk-users] asterisk as POC(push to talk) server?
AMARDEEP SINGH
- [asterisk-users] Polycom dhcp boot
colin mcdermott
- [asterisk-users] username mismatch with 1.6.2.11
Jonas Kellens
- [asterisk-users] Moving from DSL to T1
Richard Stuppi
- [asterisk-users] First boot asterisk -vvvvvgcn segfaults
Tom Lohmuller
- [asterisk-users] Synway cards
Anita Hall
- [asterisk-users] doing dnsmgr_lookup
Jonas Kellens
- [asterisk-users] Correct queue agi syntax in 1.6.2.11
Jonas Kellens
- [asterisk-users] How to send SMS to Gigaset phones ?
Olivier
- [asterisk-users] Which voicemail file format is the most widely understood ?
Olivier
- [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtimemysql
Andrew Thomas
- [asterisk-users] High volume BLF - Suggestions?
Andrew Thomas
- [asterisk-users] Force ip disconnect after register?
Bryant Zimmerman
- [asterisk-users] High volume BLF - Suggestions?
Bryant Zimmerman
- [asterisk-users] PostgreSQL is asterisk friendly with it?
Bryant Zimmerman
- [asterisk-users] High volume BLF - Suggestions?
Cassius Smith
- [asterisk-users] Speech To Text on linux with asterisk
DHAVAL INDRODIYA
- [asterisk-users] OT - Gigaset C470IP - How to access SMS settings
Olivier
- [asterisk-users] 3xx redirect response list Noop and capture
Adrian Estrada
- [asterisk-users] can asterisk accept "anonymous" register ?
zhou tianjun
- [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11
Jonas Kellens
- [asterisk-users] Random File Name
Dan Journo
- [asterisk-users] question on asterisk 1.8 meetme
Jerry Geis
- [asterisk-users] sip show channels
Dan Journo
- [asterisk-users] Random File Name
Zeeshan Zakaria
- [asterisk-users] sip show channels
Zeeshan Zakaria
- [asterisk-users] DTMF
Dan Journo
- [asterisk-users] conf checkout
Danny Nicholas
- [asterisk-users] How different is implementing Cisco based system than Asterisk based system?
Zeeshan Zakaria
- [asterisk-users] Skip Busy Agents/Channels from Queue
Shariq Khan
- [asterisk-users] One way audio when overlapdial is set to yes
leonimar cape
- [asterisk-users] Help me Out!!!!
Rob Fugina
- [asterisk-users] setting up phones
Gopalakrishnan A.N
- [asterisk-users] incoming call FXO
Zeeshan Zakaria
- [asterisk-users] Help me Out!!!!
Cassius Smith
- [asterisk-users] Dual WAN with load balancing
asterisk asterisk
- [asterisk-users] changing from zap to DAHDI
Jerry Geis
- [asterisk-users] Error loading skype_for_asterisk
Richard Kenner
- [asterisk-users] Asterisk 1.4.36 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk 1.6.2.12 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk 1.6.2.12 Download
Ryan Wagoner
- [asterisk-users] Asterisk not working with Festival
Mark G. Thomas
- [asterisk-users] Asterisk 1.6.2.13 Now Available (Re-Releast of 1.6.2.12)
Asterisk Development Team
- [asterisk-users] Queue member status not changing
Justin Sherrill
- [asterisk-users] Bug with Realtime?
Dan Journo
- [asterisk-users] Bug with Realtime?
Zeeshan Zakaria
- [asterisk-users] asterisk 1.6 and BLF
Jonas Kellens
- [asterisk-users] Realtime semi-colon
Andrew Thomas
- [asterisk-users] How to Understand a pri intense debug span X
Danny Dias
- [asterisk-users] DTMF tones too long, for once
Justin Sherrill
- [asterisk-users] [OT-FreePBX] Outbound calls check inbound routes to see if destination is local?
Tim Nelson
- [asterisk-users] [OT-FreePBX] Outbound calls check inbound routes to see if destination is local?
Tim Nelson
- [asterisk-users] one way audio for xlite clients behind NAT
Thomas Johnson
- [asterisk-users] Deadlock rendering sip useless
Ingmar Steen
- [asterisk-users] Issue with transfer (sip)
Benoit
- [asterisk-users] Sangoma A108 PCIe 2.0
Anita Hall
- [asterisk-users] Sangoma A108 PCIe V2.0
Anita Hall
- [asterisk-users] Call restriction for particular extension
Gopalakrishnan A.N
- [asterisk-users] Realtime semi-colon
Andrew Thomas
- [asterisk-users] Initial Audio Cut off
Ujjval Karihaloo
- [asterisk-users] need help with IVR dialplan
haloha
- [asterisk-users] Asterisk 1.8 and CEL logging
Bryant Zimmerman
- [asterisk-users] do carriers detect unusual / unauthorized VoIP calling patterns?
Jeff Brower
- [asterisk-users] 3rd party app store
Dean Collins
- [asterisk-users] 5-7 second delay in connecting outgoing FXO calls
Frank Tarczynski
- [asterisk-users] quick 1.8 question on console/dsp
Jerry Geis
- [asterisk-users] externip/localnet
dotnetdub
- [asterisk-users] Audiocode Median 2000 Gateway with Asterisk ?
Olivier CALVANO
- [asterisk-users] Not able to join conference
Andrew Thomas
- [asterisk-users] Confused about notifyringing in sip.conf
Jonas Kellens
- [asterisk-users] Extension continues ringing after caller hanged up
Arie Skliarouk
- [asterisk-users] Playing Audio To One Channel
Jon Farmer
- [asterisk-users] Asterisk stops processing SIP UDP messages
Daniel Tryba
- [asterisk-users] Authentication best practice
Roger Burton West
- [asterisk-users] Commands needed via AMI to find callerid of inbound call to extension
Gavin Henry
- [asterisk-users] Asterisk News Accepting Submissions
Matt Riddell
- [asterisk-users] func SHARED, how to use?
Dmitry Melekhov
- [asterisk-users] func SHARED, how to use?
Philipp von Klitzing
- [asterisk-users] func SHARED, how to use?
Dmitry Melekhov
- [asterisk-users] func SHARED, how to use?
Philipp von Klitzing
- [asterisk-users] func SHARED, how to use?
Dmitry Melekhov
- [asterisk-users] func SHARED, how to use?
Andrea Cristofanini
- [asterisk-users] func SHARED, how to use?
Dmitry Melekhov
- [asterisk-users] func SHARED, how to use?
Philipp von Klitzing
- [asterisk-users] func SHARED, how to use?
Dmitry Melekhov
- [asterisk-users] func SHARED, how to use?
Philipp von Klitzing
- [asterisk-users] func SHARED, how to use?
Dmitry Melekhov
- [asterisk-users] func SHARED, how to use?
Philipp von Klitzing
- [asterisk-users] func SHARED, how to use?
Dmitry Melekhov
- [asterisk-users] func SHARED, how to use?
Dmitry Melekhov
- [asterisk-users] func SHARED, how to use?
Philipp von Klitzing
- [asterisk-users] func SHARED, how to use?
Dmitry Melekhov
- [asterisk-users] func SHARED, how to use?
Philipp von Klitzing
- [asterisk-users] func SHARED, how to use?
Dmitry Melekhov
- [asterisk-users] func SHARED, how to use?
Dmitry Melekhov
- [asterisk-users] Dialplan extension pattern matching for '/' character
RAJNIKANT VANZA
- [asterisk-users] Not able to join conference
Andrew Thomas
- [asterisk-users] Unexplained message in 1.6.2
CDR
- [asterisk-users] digits in chan_dahdi
Marcus Vinicius
- [asterisk-users] Mixing ISDN and R2 in the same card...
Carlos Chavez
- [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
bruce bruce
- [asterisk-users] Unable to open vm-INBOXs
Jonas Kellens
- [asterisk-users] T38 and codecs negotiation
federico cabiddu
- [asterisk-users] Can't cross compile asterisk 1.6.2.13 on arm using ltib
IMS
- [asterisk-users] Asterisk T38
Adam Moffett
- [asterisk-users] TLS re-negotiation attack on SIP/TLS of Asterisk?
Fabio Pietrosanti (naif)
- [asterisk-users] http://www.asterisk.org/downloads naming schema
Klaus Darilion
- [asterisk-users] Asterisk as a distributed paging system
Matteo Fortini
- [asterisk-users] Asterisk- speech to text(Voicemail to text message)
amit salunkhe
- [asterisk-users] Installing Asterisk + FreePBX from Repsitory spits out some warnings and errors for ever
bruce bruce
- [asterisk-users] Calls stuck in the queue even when ext's are available
das sandesh
- [asterisk-users] Can't cross compile asterisk 1.6.2.13 on arm using ltib
IMS
- [asterisk-users] CDR display in minute
Mickael MONSIEUR
- [asterisk-users] realm: security issue
bilal ghayyad
- [asterisk-users] Net2Phone SIP trunk problem
Alejandro Cabrera Obed
- [asterisk-users] Can't turn debug on in a 1.2 box
khalid touati
- [asterisk-users] Sip from ip address
Geraint Lee
- [asterisk-users] realm: security issue
Zeeshan Zakaria
- [asterisk-users] Asterisk and Digium TC400B
Bryant Zimmerman
- [asterisk-users] Forking a call
Mike
- [asterisk-users] Asterisk Transfer/call patching support
Dan Cropp
- [asterisk-users] Asterisk and Digium TC400B
Bryant Zimmerman
- [asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy? (bruce bruce)
Dave Platt
- [asterisk-users] Asterisk 1.8.0 Release Candidate 2 Now Available
Asterisk Development Team
- [asterisk-users] realm: security issue
bilal ghayyad
- [asterisk-users] rtp problem with 1.8.0-rdc1
covici at ccs.covici.com
- [asterisk-users] Asterisk 1.6.2.13 - have asterisk reply from same IP address
Mike
- [asterisk-users] Redirecting a Channel more than three times...
Yves A.
- [asterisk-users] differential billing
Abdul Basit
- [asterisk-users] Debug compile fails
Daniel Tryba
- [asterisk-users] should trixbox system hang when ISP drops connection?
Robert P. J. Day
- [asterisk-users] should trixbox system hang when ISP drops connection?
Zeeshan Zakaria
- [asterisk-users] best format for playback/generation
Danny Nicholas
- [asterisk-users] best format for playback/generation
Zeeshan Zakaria
- [asterisk-users] Losing local SIP phones when internet goes down?
Gopalakrishnan A.N
- [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes
Mike
- [asterisk-users] can call internal branch , but can not call external numbers with avaya , always get return message : Q931IncompatibleDestination
Thomas Liu
- [asterisk-users] Asterisk Cluster Scenario
Stefano Sasso
- [asterisk-users] Asterisk Redundancy
Dan Journo
- [asterisk-users] 1.4, 1.6, 1.8 versions and the H323 gatekeeper functionality
bilal ghayyad
- [asterisk-users] Downloading the Asterisk as tar.gz file
bilal ghayyad
- [asterisk-users] Need to pick your brain for recommendation on using 2.5" or 3.5" HDDs for Asterisk server...
bruce bruce
- [asterisk-users] misc newbie VoIP questions
Rogelio
- [asterisk-users] PSTN to SMS and SMS to PSTN
Mian Asif
- [asterisk-users] RFC3329 support in Asterisk
sijan ahamed
- [asterisk-users] groupcount - show usage
marek cervenka
- [asterisk-users] propagate sip reinvites with directrtpsetup=yes
Eugene Oden
- [asterisk-users] Asterisk 1.62.13 - CPU spikes every 10 minutes
Dave Platt
- [asterisk-users] SCCP (skinny) phone behind NAT: RTP dest addr wrong
Infra
- [asterisk-users] How to pick a codec on the fly
Danny Nicholas
- [asterisk-users] NAT issue (i think?)
Ron
- [asterisk-users] SCCP (skinny) phone behind NAT: RTP dest addr wrong
Infra
- [asterisk-users] AstLinux 0.7.3 released
Darrick Hartman
- [asterisk-users] 1.4, 1.6, 1.8 versions and the H323 Gatekeeper functionality
bilal ghayyad
- [asterisk-users] 1.6 and 1.8 version & A2Billing
bilal ghayyad
- [asterisk-users] Cisco 9971
Damian Turburville
- [asterisk-users] ISDN - Busy signal on 3rd call
Paulo Santos
- [asterisk-users] Inbound calls from TRUNK
Khaled W. Chehab
- [asterisk-users] What's the meaning of this?
Danny Dias
- [asterisk-users] SIP X.25
Daviramos Roussenq Fortunato
- [asterisk-users] SIP X.25
Daviramos Roussenq Fortunato
- [asterisk-users] TELUS British Columbia PRI Settings
Jeremy.Hellstrom at synovate.com
- [asterisk-users] ISDN - Busy signal on 3rd call
Paulo Santos
- [asterisk-users] Weird Behavior with DAHDI
Danny Dias
- [asterisk-users] Alert-Info advice
Ishfaq Malik
- [asterisk-users] asterisk > cisco gateway > westell > isdx
Damian Turburville
- [asterisk-users] Weird Behavior with DAHDI
Andrew Thomas
- [asterisk-users] DAHDI FXO port only recognizes the "S" extension?
Andrew Thomas
- [asterisk-users] DAHDI FXO port only recognizes the "S" extension
A J Stiles
- [asterisk-users] Successive Dial apps give hang up within 30s!!
khalid touati
- [asterisk-users] can't get libpri/PRI to work, missing PRI commands
mishka at efro.us
- [asterisk-users] SCCP (skinny) phone behind NAT: RTP dest addr wrong
Infra
- [asterisk-users] Go from *100* to just 100
Jonas Kellens
- [asterisk-users] SIP Registrations
Alexandru Oniciuc
- [asterisk-users] Go from *100* to just 100
Andrew Thomas
- [asterisk-users] Asterisk 1.6.2.10 Internal timing
Jonas Kellens
- [asterisk-users] Kernel Panic When restarting the server
A J Stiles
- [asterisk-users] Asterisk 1.6.2.13 Audio Prompts Stopping
Bryant Zimmerman
- [asterisk-users] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI
Захаров Антон
- [asterisk-users] Unscheduled service outage for various Asterisk community services
Asterisk Development Team
- [asterisk-users] Intercom with Dial() works, but not with Page()
Jonas Kellens
- [asterisk-users] Unable to load fax modules
khalid touati
- [asterisk-users] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available
Ishfaq Malik
- [asterisk-users] Same extension on multiple servers confusion
Matteo Fortini
Last message date:
Thu Sep 30 15:44:31 CDT 2010
Archived on: Thu Sep 30 15:44:40 CDT 2010
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