January 2010 Archives by thread
Starting: Fri Jan 1 06:01:25 CST 2010
Ending: Sun Jan 31 23:44:34 CST 2010
Messages: 1282
- [asterisk-users] PBX Extension Help
Saeed Akhtar
- [asterisk-users] Happy New year 2010
Patterson
- [asterisk-users] Monitoring SIP & Skype connections
Administrator TOOTAI
- [asterisk-users] SIP Listen Multiple Ports
Shariq Khan
- [asterisk-users] AudioCodes MP-114 2xFXS/2xFXO - FXO not working correctly
Joseph
- [asterisk-users] Inquiry:Asterisk different codec schemes?
hadi motamedi
- [asterisk-users] Inquiry:Asterisk sip ?
hadi motamedi
- [asterisk-users] verifying correct loading of VPMADT032
ramadasan at amachu.net
- [asterisk-users] Help getting info from caller
Landy Landy
- [asterisk-users] Daily Thousands of files in recording calls in Device mode
Lenz Emilitri
- [asterisk-users] Random crashes on Bridgeaction
Markus Weiler
- [asterisk-users] Daily Thousands of files in recording calls in Device mode
Yuval Yogev
- [asterisk-users] Inquiry:How to join Asterisk real time chat?
hadi motamedi
- [asterisk-users] asterisk-users Digest, Vol 66, Issue 4
J Smith Thomas
- [asterisk-users] Outgoing Calls Only -- Firewall Rules
Nicholas Blasgen
- [asterisk-users] Dahdi causes panic on server restart
Joseph L. Casale
- [asterisk-users] Free FaxForAsterisk ReceiveFAX not working
srinivas Antarvedi
- [asterisk-users] differences between asterisk 1.6.1.x and 1.6.0.x
Giedrius Augys
- [asterisk-users] DNS reload on trunks for outgoing calls
Remco Barendse
- [asterisk-users] Some minor configuration issues with queues
jonas kellens
- [asterisk-users] Realtime Queue Members Not Ringing
Robert Broyles
- [asterisk-users] Dahdi and oslec
Joseph L. Casale
- [asterisk-users] caller getting cut off intermittently
John Taylor
- [asterisk-users] Multiple Digium cards with one NFAS trunkgroup
lesly dorval
- [asterisk-users] Script to show asterisk stuff
Tiago Geada
- [asterisk-users] H323 Disconnects after 15+ minutes
hin lee
- [asterisk-users] Register sip FXO per gateway
Joseph
- [asterisk-users] SIP Listen Multiple Ports
Vikram Ragukumar
- [asterisk-users] Dialout from Meetme conference
shrikant.soni at globussoft.com
- [asterisk-users] SIP Incoming / Inbound not working for Broadvoice (Asterisk PBX 1.6.1.6)
Qurba Joog
- [asterisk-users] lpc10
Jerry Geis
- [asterisk-users] cheap ip phone with auto-answer
Matt Riddell
- [asterisk-users] T.38 ITSP?
Karl Fife
- [asterisk-users] AGI and embargeability
Quinn Weaver
- [asterisk-users] Inquiry:Asterisk sending dialed digits in one-by-one digit format?
hadi motamedi
- [asterisk-users] automatic dial from database
shameem Banu
- [asterisk-users] {Spam?} MeetMe/Dahdi for FreeBSD
Leif Neland
- [asterisk-users] Get Queue Info
Daniel Stefanus
- [asterisk-users] Realtime LDAP Queues crashes
Jorge Salamero Sanz
- [asterisk-users] CallerID on Indian PSTN is not working.
Arun Sasidhar
- [asterisk-users] Dialplans & Holiday Dates
Danny Nicholas
- [asterisk-users] DTMF detection on dahdi with b4xxp (again, some more details)
Christian Theune
- [asterisk-users] Canadian call quality issue
Max McGraw
- [asterisk-users] Really Silly Question From Total Newbie
UIT DEVELOPMENT
- [asterisk-users] [asterisk-speech-rec] AGI and embargeability
Quinn Weaver
- [asterisk-users] send faxes as "3,1 kHz Audio"
achris at abacus6.net
- [asterisk-users] Merlin Legend integration not routing calls back to PSTN.
Shane Brath
- [asterisk-users] Originate from the Dialplan
Matthew Edmondson
- [asterisk-users] Fastagi-mapping problem
ahmed magdy
- [asterisk-users] Inquiry:How to define incoming route for sip?
hadi motamedi
- [asterisk-users] Zaptel compilation problems: Data Mode!!
mosleh at infolog.mr
- [asterisk-users] Asterisk 1.6.1.x SMDI MWI w/Fujitsu F9600 Problem
Will Szopko
- [asterisk-users] question on makefile
Jerry Geis
- [asterisk-users] DEVICE STATE "In use"
Tiago Geada
- [asterisk-users] Urgent: Which spandsp version is recommended for 1.6.1 ?
Olivier
- [asterisk-users] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P
Olivier
- [asterisk-users] video with x-lite
Jerry Geis
- [asterisk-users] How to see STDERR message?
Zhang Shukun
- [asterisk-users] Question about PLC of Asterisk
nakaji at 02.246.ne.jp
- [asterisk-users] compile one additional module without recompiling all asterisk
Giedrius Augys
- [asterisk-users] error compile dahdi with latest kernels.
james.zhu
- [asterisk-users] Explain what asterisk.conf's "internal timing" option is
Olivier
- [asterisk-users] queue and linear strategy
Giedrius Augys
- [asterisk-users] Zaptel compilation problems: Data Mode!!
mosleh at infolog.mr
- [asterisk-users] How to dial a number make two phone Ring at the same time?
Zhang Shukun
- [asterisk-users] Dialing OutBound SIP trunk using Dial() command
srinivas Antarvedi
- [asterisk-users] Please remove me from the mailing list.
Rick Dean
- [asterisk-users] Sip REFER failes w/603 Decline (Policy), Polycom Phone
Steve Totaro
- [asterisk-users] Crash in Asterisk
Danny Dias
- [asterisk-users] AGI perl script set timeout within script?
JR Richardson
- [asterisk-users] AGI perl script set timeout within script?
JR Richardson
- [asterisk-users] How to recieve number returned by $AGI->wait_for_digit($timeout)
Zhang Shukun
- [asterisk-users] Faxing: Anyone have a compiled executable?
David Backeberg
- [asterisk-users] Cheap femtocell's ahead
Jay R. Worthington
- [asterisk-users] [VUC] Today at 12 Noon EST (6PM CEST, 9AM PST) iNum with Voxbone
Randy R
- [asterisk-users] Semi-OT: Configuring SIP trunks with Cisco UCM 7.0.
Kristian Kielhofner
- [asterisk-users] Queue_log file and mysql logs together!
Dpto. de Sistemas
- [asterisk-users] Faxing: Anyone have a compiled executable?
Kristijan Vrban
- [asterisk-users] Multicast RTP Paging
Krishna Sumanth Chava
- [asterisk-users] How can I get codec info on active calls
Landy Landy
- [asterisk-users] AGI perl script set timeout within script?
JR Richardson
- [asterisk-users] Asterisk CallerId problem?
hadi motamedi
- [asterisk-users] Choppy MOH
--[ UxBoD ]--
- [asterisk-users] Quick Installing Asterisk-1.4 on Ubuntu
Valter Nogueira
- [asterisk-users] UK dialing tone
--[ UxBoD ]--
- [asterisk-users] Using HASH() and REALTIME_HASH()
Benoit
- [asterisk-users] Queue - Update CDR
Marcelo
- [asterisk-users] Music / Background
Thomas Perron
- [asterisk-users] No dial-tone with X101P FXO card
Nitin Bahadur
- [asterisk-users] You won't help me anymore?
hadi motamedi
- [asterisk-users] Off-line subscribed phone amber on SPA942?
Leif Neland
- [asterisk-users] Directory and Voicemail Problems after upgrading from 1.4 to 1.6
Christopher Wolff
- [asterisk-users] Grandstream GXW-4024
C F
- [asterisk-users] app_swift 1.6.2 DTMF issue
Jeremy Kister
- [asterisk-users] Zhang Shukun 想跟您聊天
Zhang Shukun
- [asterisk-users] How to use AGI php script function $agi -> exec_dial
Zhang Shukun
- [asterisk-users] How to test if a telephone is busy now?
Zhang Shukun
- [asterisk-users] asterisk-users archive
Jeremy Kister
- [asterisk-users] Attempted break in ?
--[ UxBoD ]--
- [asterisk-users] Temporary loss of audio on all SIP channels
Tony Mountifield
- [asterisk-users] Extension Status
ahmed magdy
- [asterisk-users] Skype for Asterisk
A.Santoro
- [asterisk-users] TONIGHT Join 5-6P Mon 11th - 1st Evening Meeting test IRC & VOIP online Asterisk at BerkeleyTIP-Global - for forwarding
giovanni_re
- [asterisk-users] Asterisk core dumps when using PrivacyManager
--[ UxBoD ]--
- [asterisk-users] Sipgate > DTMF not detected
listuser at spamomania.co.uk
- [asterisk-users] Custom date formats with new mode say.conf?
Murray Melvin
- [asterisk-users] ChanSpy doesn't hangs up
Joao Gomes Pereira
- [asterisk-users] SIP over VPN -- no audio to other remote/VPN connected phones
Ryan McCormack
- [asterisk-users] Problem with call transfer and Polycom 430
Mike Diehl
- [asterisk-users] Interfacing to NEC Xen Master PBX
John Treen
- [asterisk-users] is roundrobin and rrmemory the same meaning?
Zhang Shukun
- [asterisk-users] Why agent log out automaticly?
Zhang Shukun
- [asterisk-users] Beginners Guide to setting up a Call Centre
Peter Childs
- [asterisk-users] Virtual ISDN device /dev/XYZ
Roger Schreiter
- [asterisk-users] Send 503 or 603 error after answer()
jonas kellens
- [asterisk-users] Question about SIP registration
Aggio Alberto
- [asterisk-users] Inserting a wait in a sip dial
evert at disruptor.nl
- [asterisk-users] Minimal Asterisk Web Interface?
Tim Nelson
- [asterisk-users] VMs & IMAP Storage
--[ UxBoD ]--
- [asterisk-users] Problem logs queue_log-mysql
Dpto. de Sistemas
- [asterisk-users] AudioCodes MP-114 - SAS (Stand Alone Survivability) configuration
Joseph
- [asterisk-users] Xorcom 32 channel FXS gateway
C F
- [asterisk-users] Odd Voicemail Issue
William Stillwell ( Lists )
- [asterisk-users] Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
hadi motamedi
- [asterisk-users] CallerID on Indian PSTN is not working.
Arun Sasidhar
- [asterisk-users] FW: [mythtv-users] VMWare on the backend. Viable solution?
Dean Collins
- [asterisk-users] Asterisk 1.4.28 intermittent one way audio?
JR Richardson
- [asterisk-users] asterisk / NEC2400 / PRI
Anthony Geoffron
- [asterisk-users] is there some Chinese version of sounds available?
Zhang Shukun
- [asterisk-users] ISDN Cause codes for unanswered calls
Steve Moran
- [asterisk-users] Attend CampKDE Jan 15-18 via Voice over Internet (VOIP), BerkeleyTIP
giovanni_re
- [asterisk-users] What about the performance visit MYSQL in DialPlan code?
Zhang Shukun
- [asterisk-users] how to strip + from the caller-ID
Szasz Szabolcs
- [asterisk-users] Lagged Extension
--[ UxBoD ]--
- [asterisk-users] different between ring groups and queue?
Zhang Shukun
- [asterisk-users] Followme Options
Positively Optimistic
- [asterisk-users] Followme Options
Positively Optimistic
- [asterisk-users] iaxmodem / hylafax receive problem
Kingsley Tart
- [asterisk-users] Dahdi issues
Jeff LaCoursiere
- [asterisk-users] Languages
Örn Arnarson
- [asterisk-users] Can not play WAV-files attached to mail from my own Asterisk
jonas kellens
- [asterisk-users] Ringing for incoming call
Andrew Thomas
- [asterisk-users] Dahdi and FreePBX
Jeff LaCoursiere
- [asterisk-users] Problem logs queue_log-mysql
Dpto. de Sistemas
- [asterisk-users] Fax Detection on SIP
--[ UxBoD ]--
- [asterisk-users] GXV3140 and Xlite video
Julian Lyndon-Smith
- [asterisk-users] Friday Jan 15 @12 Noon EST: Hacking VoIP
Randy R
- [asterisk-users] 10/100 voip phones and gigabit connection
randall
- [asterisk-users] Question about Presence and IM feature
Yuji Kondo
- [asterisk-users] Realtime queue not work
Zhang Shukun
- [asterisk-users] jitterbuffer and PLC
nakaji at 02.246.ne.jp
- [asterisk-users] OT: Inbound South America numbers
Administrator TOOTAI
- [asterisk-users] Logs problem of queue_log-mysql
Dpto. de Sistemas
- [asterisk-users] : Asterik with out registration.
Aditya Kumar
- [asterisk-users] Asterisk 1.4.29 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk 1.6.0.21 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk 1.6.1.13 Now Available
Asterisk Development Team
- [asterisk-users] Asterisk 1.6.2.1 Now Available
Asterisk Development Team
- [asterisk-users] DAHDI and Analogue lines (UK)
Gordon Henderson
- [asterisk-users] Changing ring cadence on FXS lines
Noah I. Engelberth
- [asterisk-users] info for Busy for incoming internal call but not for exterrnal
lore
- [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
Bruce Nik
- [asterisk-users] TE410P generates only 1 interrupt
Srinath
- [asterisk-users] Echo on Polycom phones
hin lee
- [asterisk-users] Realtime cached values
Deep D
- [asterisk-users] Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
Zhang Shukun
- [asterisk-users] Howto regret blind transfer?
hbk
- [asterisk-users] Hint for realtime peers
Deep D
- [asterisk-users] Cross compiling Asterisk, Dahdi..
Gordon Henderson
- [asterisk-users] Do any providers support speex codec?
Daniel Clark
- [asterisk-users] How to escape the Pound-Char in Callfiles
Dominik
- [asterisk-users] How to escape characters in Dialplan
Dominik
- [asterisk-users] receive text
Thomas Perron
- [asterisk-users] Dial String command after audio background
Thomas Perron
- [asterisk-users] help with picking out a digium card.
shawn bright
- [asterisk-users] Will SIP connection stop automaticlly when detect no voice between the channel after a period of time?
Zhang Shukun
- [asterisk-users] What's customer_id mean?
Zhang Shukun
- [asterisk-users] How to play the voicemail recorded?
Zhang Shukun
- [asterisk-users] Dahdi/callerid issue
evert at disruptor.nl
- [asterisk-users] TDM2400P "Unable to set SW Companding on channel .."
wassim darwich
- [asterisk-users] Asterisk answers with wrong sip entry
Joseph
- [asterisk-users] wav to gsm can't play
Zhang Shukun
- [asterisk-users] How to retrieve a phone number fromcall forwarding?
Soonthorn Ativanichayaphong
- [asterisk-users] B2bua
ahmed magdy
- [asterisk-users] test case with queues and system()
Евгений Шишкин
- [asterisk-users] Call drop-out on second incoming call.
Mike Diehl
- [asterisk-users] Initialize mailbox greeting message
Olivier
- [asterisk-users] How to enable a range of IP addresses in realtime sip_buddies
Bruce Ferrell
- [asterisk-users] Which to choose? Realtime extension OR Static extension with MYSQL command
Zhang Shukun
- [asterisk-users] Selecting IP address for RTP
Richard Kenner
- [asterisk-users] Odd message: "correct auth, but ..."
Richard Kenner
- [asterisk-users] Polycom Soundstation Conferencing Unit
Dean Collins
- [asterisk-users] AstLinux 0.7.0 Released
Darrick Hartman
- [asterisk-users] More than a line with same extension + Polycom 320 + Provision Tool
bilal ghayyad
- [asterisk-users] Call Xfer issue between DataCenter and User Site
Steven Davison
- [asterisk-users] More than a line with same extension + Polycom + Provision Tool
bilal ghayyad
- [asterisk-users] DTMF Issue?
hin lee
- [asterisk-users] Using SIPPEER status with CUT function?
JR Richardson
- [asterisk-users] Using SIPPEER status with CUT function? SOLVED
JR Richardson
- [asterisk-users] DAHDI-Linux 2.2.1 and DAHDI-Tools 2.2.1 Released
Asterisk Development Team
- [asterisk-users] queue groups in asterisk 1.4
Steven Alligood
- [asterisk-users] Setting MixMonitor options from Queue
Scott Gifford
- [asterisk-users] Asterisk 403 Forbidden message with port translation
Vikram Ragukumar
- [asterisk-users] Pass-through Call Recording Transfer Information
Glen Ganderton
- [asterisk-users] Asterisk & LDAP authentification
Jonathan Barou
- [asterisk-users] DTMF reception during WaitForSilence
Yves Arikoglu
- [asterisk-users] Caller hang up not detected
hugolivude
- [asterisk-users] Feature codes not detected
hugolivude
- [asterisk-users] odbc question
Giedrius Augys
- [asterisk-users] Echo cancellation in a sip channel
Alexandre Rodrigues
- [asterisk-users] chan_ss7 or libss7, which is more stable?
equis software
- [asterisk-users] pri CLI command not available
Eric Merkel (Mail Lists)
- [asterisk-users] Popular Gigabit Phones
Matt Darnell
- [asterisk-users] Trouble getting feature codes to work
hugolivude
- [asterisk-users] MYSQL problem
Zhang Shukun
- [asterisk-users] GoToIfTime issue
Zhang Shukun
- [asterisk-users] OfficeSIP Softphone
Vitali Fomine
- [asterisk-users] FW: Call Xfer issue between DataCenter and User Site
Steven Davison
- [asterisk-users] OT - SPA3102 not detecting CID - Which settings to tune ?
Olivier
- [asterisk-users] Meetme conferencing - large deployment SIP or ZAP?
Steve Moran
- [asterisk-users] Asterisk 1.6 mysql 'NO ANSWER' disposition problem
Artifex Maximus
- [asterisk-users] Snom vs Polycom
Julian Lyndon-Smith
- [asterisk-users] Set CDR userfield for Queues
Deep D
- [asterisk-users] Polycom phone DND state
Mike
- [asterisk-users] Handling SIP error codes/ISDN codes
das sandesh
- [asterisk-users] Siemens Gigaset + Asterisk Query?
Alan Lord (News)
- [asterisk-users] IAX ans SS7
mickael ropars
- [asterisk-users] Siemens Gigaset + Asterisk Query?
John Hurley
- [asterisk-users] Set CDR userfield for Queues
Deep D
- [asterisk-users] Xorcom problem after update from zaptel to dahdi-2.2.1
Loic Didelot
- [asterisk-users] fax over IP - http/ftp-provisioning - intercom
jonas kellens
- [asterisk-users] Jabber Server
ahmed magdy
- [asterisk-users] odd issue with the with SIP over VPN
Zane C.B.
- [asterisk-users] AOC advise of charge
antselva
- [asterisk-users] ReceiveFAX and SendFAX questions
Magnus Benngård
- [asterisk-users] odd issue with the with SIP over VPN
Dave Platt
- [asterisk-users] two stage dialing in a SIP dial plan
Bruce Ferrell
- [asterisk-users] [OT] Snom M3s
--[ UxBoD ]--
- [asterisk-users] Adminpin for conference room
Deepesh D
- [asterisk-users] MySQL RealTime Error
Zhang Shukun
- [asterisk-users] MYSQL grammar diff in 1.6.2.1?
Zhang Shukun
- [asterisk-users] Detected digit 'f'
Kingsley Tart
- [asterisk-users] Web-Meetme 4.0 and Asterisk 1.6.2
joern
- [asterisk-users] queue
bhrugu mehta
- [asterisk-users] sip.conf with versatel and two NICs very strange problem
Yves Arikoglu
- [asterisk-users] Call recordings and sensitive information
Julian Lyndon-Smith
- [asterisk-users] Call tagging
Julian Lyndon-Smith
- [asterisk-users] ASTSBINDIR not being picked up by safe_asterisk
Mark Hulber
- [asterisk-users] [OT] spa3000 (Regional & Line1) NL settings required
pepesz
- [asterisk-users] How to make SpeechBackground keep playing if utterance doesn't match our grammar
Quinn Weaver
- [asterisk-users] Disa not fully bridging outbound call
John Millican
- [asterisk-users] StopPlayTones() after first digit?
Jack Bates
- [asterisk-users] Asterisk 1.2.37 + BLF + ParkedCalls + SPA962
Joel Lansden
- [asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops
Peter Childs
- [asterisk-users] Error and call drops
Lee Archer
- [asterisk-users] settings for soft phones
Eric Smith
- [asterisk-users] [inter-pbx commnication] trying to make PBX1 talk to PBX2
khalid touati
- [asterisk-users] Anyone going to HD Communications Summit - Europe Feb 12th?
Randy R
- [asterisk-users] Attended Transfer with REFER
Örn Arnarson
- [asterisk-users] SIP Hard Phone with SMS
"Juan E. Rodríguez"
- [asterisk-users] Realtime Queue not work in 1.6.2.1
Zhang Shukun
- [asterisk-users] CDR messed up when using queue
jonas kellens
- [asterisk-users] Connecting to an External EPBX without an SIP provider
Siju George
- [asterisk-users] Unregistred users can pass calls, peer being static
Administrator TOOTAI
- [asterisk-users] astdb
bhrugu mehta
- [asterisk-users] Asterisk, NAT, and RTP?
Vincent
- [asterisk-users] CDR problems with Queue
Håkon Nessjøen
- [asterisk-users] Need recommendation for ISDN-BRI cards for use with Asterisk
Zeeshan Zakaria
- [asterisk-users] Mitel integration
Jeff LaCoursiere
- [asterisk-users] Asterisk Database Configuration
ahmed magdy
- [asterisk-users] PRI Connected to definity errors
Alec Davis
- [asterisk-users] yum install asterisk16 for Fedora Core 8
Bruce Nik
- [asterisk-users] Linux-based hard phones?
Ken D'Ambrosio
- [asterisk-users] Database Configration
ahmed magdy
- [asterisk-users] iax softphones - not reconnecting
Asterisk - thinking:systems
- [asterisk-users] iax client for symbian s60
Asterisk - thinking:systems
- [asterisk-users] Asterisk Database
ahmed magdy
- [asterisk-users] Inserting white noise / music / sound file into mixmonitor
Julian Lyndon-Smith
- [asterisk-users] Fw: OfficeSIP Softphone
Vitali Fomine
- [asterisk-users] Polycom Soundpoint 300IP
jonas kellens
- [asterisk-users] New feature in app_queue: Give members a penalty time for not answering (help testing)
Håkon Nessjøen
- [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short
wassim darwich
- [asterisk-users] 911, location
mir shahnawaz
- [asterisk-users] 911, location
Barry L. Kline
- [asterisk-users] 911, location
mir shahnawaz
- [asterisk-users] 911, location
Danny Nicholas
- [asterisk-users] 911, location
mir shahnawaz
- [asterisk-users] 911, location
Doug Lytle
- [asterisk-users] 911, location
--[ UxBoD ]--
- [asterisk-users] 911, location
Leif Neland
- [asterisk-users] 911, location
Danny Nicholas
- [asterisk-users] 911, location
Leif Neland
- [asterisk-users] 911, location
Danny Nicholas
- [asterisk-users] 911, location
Kevin P. Fleming
- [asterisk-users] 911, location
Shahnawaz Mir
- [asterisk-users] 911, location
Kyle Kienapfel
- [asterisk-users] TDM2400 card FXS problems
Noah I. Engelberth
- [asterisk-users] [inter-pbx commnication] trying to make PBX1 talk to PBX2
khalid touati
- [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it possible?
khalid touati
- [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short
wassim darwich
- [asterisk-users] Use of "603 Declined"
Kristian Kielhofner
- [asterisk-users] AsyncGoto/DAHDI ?
hin lee
- [asterisk-users] How to set sip client idle or busy in Asterisk ?
Allway
- [asterisk-users] Cell Phone dialing
Danny Nicholas
- [asterisk-users] beroNet BN4S0e PCI Express ISDN Card with chan_dahdi
Laurent CARON
- [asterisk-users] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
Walter Arguello
- [asterisk-users] chan_mobile problem with audio (distorted)
Marian Zahariev
- [asterisk-users] VUC Today at 1 PM EST: Counterpath/Bria
Randy R
- [asterisk-users] Address family not supported by protocol
Chris Gentle
- [asterisk-users] disable comfort noise
Szasz Szabolcs
- [asterisk-users] 1 Asterisk server, multiple registrations to ITSP
jonas kellens
- [asterisk-users] TDM2400 card FXS problems
Noah I. Engelberth
- [asterisk-users] Cell phone redialer?
Myles Wakeham
- [asterisk-users] Problem with ringing (or absence of...) with Polycom forwarding
Mike
- [asterisk-users] TDM2400 card FXS problems
garry liu
- [asterisk-users] Dial cellphone from one PBX1 to PBX2? is it possible?
khalid touati
- [asterisk-users] smsq command
Jerry Geis
- [asterisk-users] Questions about asterisk and spa2102
Kosa
- [asterisk-users] microphone on Polycom 550/650
hin lee
- [asterisk-users] Help configuring Audiocodes MP-104 FXO
Daniel - Asterisk
- [asterisk-users] New feature: Asterisk Manager Interface commands for DeviceState
Håkon Nessjøen
- [asterisk-users] Broker lines on a T1 : Signaling convention?
Martin Andrews
- [asterisk-users] Digium fax - sending fax call file vs manager originate
Hristo Benev
- [asterisk-users] Help for MOH - sounding scratchy/static on hold
das sandesh
- [asterisk-users] callerid not working over sip
sean darcy
- [asterisk-users] Caller ID not working properly on some phones...
Carlos Chavez
- [asterisk-users] Asterisk status "488 Not acceptable here" on receiving fax
Deepesh D
- [asterisk-users] forward call back up same trunk to external cell phone problem
John Taylor
- [asterisk-users] Aastra RFP-32 and CLID
Magnus Benngård
- [asterisk-users] Set CDR userfield for Queues
Luis Morales
- [asterisk-users] MATH
Thomas Perron
- [asterisk-users] FAX over ISDN PRI
Mariano Lecuona
- [asterisk-users] microphone on Polycom 550/650
Michael Graves
- [asterisk-users] MATH
Thomas Perron
- [asterisk-users] asterisk-users Digest, Vol 66, Issue 75
Muro, Sam
- [asterisk-users] 911, Location
Muro, Sam
- [asterisk-users] sip to dahdi and billsec
Uros Djokic
- [asterisk-users] request for testing: MixMonitor Mute
Julian Lyndon-Smith
- [asterisk-users] SIP Registration Failure Logging
Jim Rosenberg
- [asterisk-users] app_directory broken in 1.6
cjwstudios
- [asterisk-users] SIP Registration Failure Logging
uzzi
- [asterisk-users] Odd error mssage on DAHDI lines
Richard Kenner
- [asterisk-users] FCT for 3G Video calls
Chris Hills
Last message date:
Sun Jan 31 23:44:34 CST 2010
Archived on: Sun Jan 31 23:50:10 CST 2010
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