January 2007 Archives by thread
Starting: Mon Jan 1 07:36:06 MST 2007
Ending: Wed Jan 31 23:53:22 MST 2007
Messages: 2514
- [asterisk-users] Thomson ST2020 and voicemail
Dante Dante
- [asterisk-users] X100P "rings" randomly when "phone" line makes
call
Zoilo Gomez
- [asterisk-users] Help needed with Polycom dialplan pattern matching
John French
- [asterisk-users] Happy 2007!!!
Bill Hackensack
- [asterisk-users] How to connect two asterisk server
sunil at koltelecom.com
- [asterisk-users] (OT) Where to post free source for AGI?
Lee Jenkins
- [asterisk-users] Re: Hi reg. 2 asterisk server
Thirumal Saminathan
- [asterisk-users] Problem with centos 4.4 and jabber/gtalk (really
iksemel)
Kenneth Padgett
- [asterisk-users] Dialed Number missing from the CDR
when usingcallfiles.
Michael Collins
- [asterisk-users] Dual Ringing Tones
Florian Overkamp
- [asterisk-users] chan_oh323 early media
Jason Kim
- [asterisk-users] asterisk and mysql
RdBSD
- [asterisk-users] Realtime multiple registration for a Hard Phone
Snom 360
Olivier
- [asterisk-users] asterisk and mysql
Ngo Duc Loi
- [asterisk-users] [OT] Wifi SIP phones - LinkSys WIP330
Olivier
- [Asterisk-Users] asterisk + door opener
Terry Wade
- [asterisk-users] Avoiding deadlock-line drop problem
Giannis Margaritis
- [asterisk-users] Best Hardware for Asterisk Server?
Mark Greene
- [asterisk-users] Save SIP DEBUG output to a file
Frederico Madeira
- [asterisk-users] PRI ANI/CallerID
Jerry Jones
- [asterisk-users] [OT] Wifi SIP phones - LinkSys WIP330
Jorge Mendoza
- [asterisk-users] asterisk and mysql
Savoy, Kevin - Williston, ND
- [asterisk-users] [asterisk-biz] Slightly updated UK English voice
prompts
Steve Kennedy
- [asterisk-users] (OT) Where to post free source for AGI?
Ejay Hire
- [asterisk-users] 802.1x support in wired sip hardphones ?
Olivier
- [asterisk-users] (OT) Where to post free source for AGI?
Bruce Reeves
- [asterisk-users] (OT) Where to post free source for AGI?
Lenz
- [asterisk-users] (OT) Where to post free source for AGI?
Tzafrir Cohen
- [asterisk-users] How to show a debugging remark in a sip or
extensions context?
Larry Alkoff
- [asterisk-users] (OT) Where to post free source for AGI?
Michael Collins
- [asterisk-users] Best Hardware for Asterisk Server?
joe a.
- [asterisk-users] SpanDSP and Asterisk 1.4
Mark Johnson
- [asterisk-users] Re: Grandstream GXW-4108 8 port FXO
Martin Joseph
- [asterisk-users] yet another faxing issue (outbound only, via ATA)
Bill Gibbs
- [asterisk-users] Call connected, cannot hear or speak - $20 for fix
zero massive
- [asterisk-users] Best Hardware for Asterisk Server?
Colin Anderson
- [asterisk-users] (OT) Where to post free source for AGI?
Lee Jenkins
- [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source
Joao Pereira
- [asterisk-users] queues - limiting ringing calls to queue members
Nikola Ciprich
- [asterisk-users] RE: yet another faxing issue (outbound only,
via ATA)
Bill Gibbs
- [asterisk-users] vzaphfc?
Remco Barendse
- [asterisk-users] OT: Admin manual for Linksys Sipura SPA-2102
Erick Perez
- [asterisk-users] OnHook Call Announcement...
Carlos Chavez
- [asterisk-users] extension problems
Vulpes Velox
- [asterisk-users] Double quotes in CDRUserField?
Michael Collins
- [asterisk-users] RE: yet another faxing issue (outbound only,
via ATA)
Bill Gibbs
- [asterisk-users] Re: [A*UG] How to show a debugging remark in a sip
or extensions context?
Larry Alkoff
- [asterisk-users] connecting asterisk (trixbox) to traditional phone
lines?
blackwater dev
- [asterisk-users] Error compiling chan_vpb
Kevin P. Fleming
- [asterisk-users] Dialed Number missing from the CDR when using
callfiles.
Michael Collins
- [asterisk-users] Re: [A*UG] How to show a debugging remark in a sip
or extensions context?
Larry Alkoff
- [asterisk-users] SNOM loses server registration
Joao Pereira
- [asterisk-users] Dubai Caller ID
Mihaly Antal
- [asterisk-users] ISA server Issue (Maybe off topic)
Mattias Andersson
- Fwd: [asterisk-users] Disconnect supervision in India?
Rajkumar S
- [asterisk-users] voice fax modem and asterisk
Gregory Machin
- [asterisk-users] native music on hold distortion between files
Damon Estep
- [asterisk-users] Sangoma Remora A202
Todd H
- [asterisk-users] MeetMe() not recording calls
John French
- [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo
/audio artifacts
Colin Anderson
- [asterisk-users] Fonebridge2
Jon Schøpzinsky
- [asterisk-users] Voicemail to email
Mark Greene
- [asterisk-users] Best Hardware for Asterisk Server?
joe a.
- [asterisk-users] Sangoma A102 w/ EC module gets intermittent
echo /audio artifacts <---More information
Colin Anderson
- [BULK] [asterisk-users] Fonebridge2
Savoy, Kevin - Williston, ND
- [asterisk-users] Sangoma A102 w/ EC module gets intermittent
echo/audio artifacts
Colin Anderson
- [asterisk-users] answer machine detection
Julian Lyndon-Smith
- [asterisk-users] SIP Dial out timeout
Arik Raffael Funke
- [asterisk-users] Polycom Power Specs
Peder at NetworkOblivion
- [asterisk-users] API: how to bridge originated call?
chester c young
- [asterisk-users] Is chan_zap.so loaded?
John French
- [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
Dan Austin
- [asterisk-users] Park and Page
Mike Clark
- [asterisk-users] Sangoma Remora A202
Pierre Marceau
- [asterisk-users] Error on answer a SIP 401 message
Frederico Madeira
- [asterisk-users] Cisco 79x1 Auto-Answer
Jeremiah Millay
- [asterisk-users] have a phone number from stanaphone and a working
trixbox, how do I connect them?
blackwater dev
- [asterisk-users] over 200 queues, anyone?
lenz
- [asterisk-users] over 200 queues, anyone?
Joe Dennick
- [asterisk-users] over 200 queues, anyone?
Alex Robar
- [asterisk-users] over 200 queues, anyone?
Joe Dennick
- [asterisk-users] over 200 queues, anyone?
Lenz
- [asterisk-users] over 200 queues, anyone?
Gavin Hamill
- [asterisk-users] over 200 queues, anyone?
lenz
- [asterisk-users] over 200 queues, anyone?
Leo Ann Boon
- [asterisk-users] over 200 queues, anyone?
Lenz
- [asterisk-users] over 200 queues, anyone?
BJ Weschke
- [asterisk-users] over 200 queues, anyone?
Leo Ann Boon
- [asterisk-users] over 200 queues, anyone?
Terry Wade
- [asterisk-users] over 200 queues, anyone?
Olivier
- [asterisk-users] over 200 queues, anyone?
Rob Lith
- [asterisk-users] over 200 queues, anyone?
Lenz
- [asterisk-users] over 200 queues, anyone?
Richard Lyman
- [asterisk-users] have a phone number from stanaphone and a
workingtrixbox, h
Yuan LIU
- [asterisk-users] ARI help
Mark Greene
- [asterisk-users] Gentoo ebuild for 1.4?
Chris Bagnall
- [asterisk-users] Detect IP path before calling
Yuan LIU
- [asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
Douglas Garstang
- [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released
Dan Austin
- [asterisk-users] Any quiet 24 port POE switches out there?
John French
- [asterisk-users] v140 ./configure not finding installed ssl
snowcrash+asterisk
- [asterisk-users] [Announce] Web-MeetMe 3.0.0 RE-released
Dan Austin
- [asterisk-users] 1.4 segfaulting when manager client is connected
Brad Templeton
- [asterisk-users] ztdummy on 1.6
chester c young
- [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
Dan Austin
- [asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 4
Edwin Groothuis
- [asterisk-users] caller id ring tones for Asterisk Phone
Jeronimo Romero
- [asterisk-users] asterisk sip peer/user matching methods for
authentication backwards?
Damon Estep
- [asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
Douglas Garstang
- [asterisk-users] Cisco AS5300
yusuf
- [asterisk-users] Required freelancer for installing hylafax on
Asterisk Box
gkanuganti at mantragroup.com
- [asterisk-users] bypass menu for certain numbers?
Matt Gibson
- [asterisk-users] Hi reg. asterisk Compilation
Thirumal Saminathan
- [asterisk-users] Maybe a NAT problem
Facundo Barrera - GMail
- [asterisk-users] Digium Wildcard B410P
Henrik Woffinden
- [asterisk-users] postgres and asterisk
O.Kamal
- [asterisk-users] Digium Wildcard B410P
Richard Soderblom
- [asterisk-users] Create a group of SIP acoount for outgoing calls ?
Noc Phibee
- [asterisk-users] Realtime voicemail passwords
Bruce Ferrell
- [asterisk-users] mISDN crypto?
Andreas Anderson
- [asterisk-users] PRI Problems
Rob Schall
- [asterisk-users] System() and Trysystem() in extensions.conf => get
the result ?
Noc Phibee
- [asterisk-users] [Fwd: PRI Problems]
Rob Schall
- [asterisk-users] over 200 queues, anyone?
lists at infoway.net
- [asterisk-users] Sangoma A102 w/ EC module gets intermittent
echo/audio artifacts <--followup and resolution
Colin Anderson
- [asterisk-users] Convert a file from WAV to WAV49 or GSM for
Asterisk
Chris Carey
- [asterisk-users] Asterisk 1.4.0 segfault
Ondrej Valousek
- [asterisk-users] asterisk sip peer/user matching methods
forauthentication backwards?
Doug Meredith
- [asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
Douglas Garstang
- [asterisk-users] #include not working in 1.4
Bruce Ferrell
- [asterisk-users] SIP peer lookup problems
Jon Schøpzinsky
- [asterisk-users] Re: Codec swap (reinvite)
Julian J. M.
- [asterisk-users] Best inexpensive home office router for VoIP (QoS
with maybe PoE)
Mike
- [asterisk-users] asterisk sip peer/user matching
methodsforauthentication backwards?
Damon Estep
- [asterisk-users] Re: Re: [Announce] Web-MeetMe 3.0.0 released
Dan Austin
- [asterisk-users] Trouble compiling asterisk 1.2.14
Guillermo Salas M.
- [asterisk-users] How big a pipe can IAX2 go?
Adrian Marsh
- [asterisk-users] Re: Alert: Steering Committee Reminder and Agenda
Flash Love
- [asterisk-users] proxy howto
Mark Price
- [asterisk-users] TE110P with Qsig
Josué Conti
- [asterisk-users] [resolved] asterisk 1,4 and google talk
Ronald Lewis
- [asterisk-users] How to routing call to Quintum.
Reaz
- [asterisk-users] Dimensioning a 50 sip phone installation
Erick Perez
- [asterisk-users] HowTO configure voice T1
Mark Greene
- [asterisk-users] DISA Ring Back
Supa
- [asterisk-users] MusicOnHold Files
Forrest Beck
- [asterisk-users] IAX vs SIP trunks between Asterisk boxes
Damon Estep
- [asterisk-users] 3-way calling MGCP capture
Olga Mill
- [asterisk-users] POE draw on Aastra 480i
Allen Casteran
- [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
Dan Austin
- [asterisk-users] Which is GUI to edit Asterisk IVR logic
Olivier
- [asterisk-users] faxing times!
Thirumal Saminathan
- [asterisk-users] chan_zap.c: Failed to read gains: Invalid argument
Lee Archer
- [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument
Lee Archer
- [asterisk-users] Which g729 module for HP DL 360 G3 (Xeon CPU's)?
Eric Bishop
- [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument
Lee Archer
- [asterisk-users] asterisk 1.4.0 didn't compile chan_zap.so
Ma Zhiyong
- [asterisk-users] Re: Best inexpensive home office router for VoIP
(QoS with maybe PoE)
Robbie Hughes
- [asterisk-users] POE draw on Aastra 480i
Watkins, Bradley
- [asterisk-users] How to build 1.4 with res_crypto.so
Yann Massard
- [asterisk-users] Invalid DivertingLegInformation2 component
received 0x38
Andreas Gaufer
- [asterisk-users] fax transmission
Vieri Di Paola
- [asterisk-users] anyone using metermaid / parked call BLF?
Dr. Michael J. Chudobiak
- [asterisk-users] Asterisk and IM
Hall, Eric M.
- [asterisk-users] integrating with Asterisk and OpenSER for Voicemail
raviprakash sunkara
- [asterisk-users] Re: [Users] integrating with Asterisk and OpenSER
for Voicemail
Steve Blair
- [asterisk-users] idle SIP channels problem
O.Kamal
- [asterisk-users] addons 1.4 and cdr_addon_mysql not installed !
Luca Lafranchi Lists
- [asterisk-users] how to transfer calls when analog phone has no
transfer button
Erick Perez
- [asterisk-users] chan_zap.c: Failed to read gains: Invalidargument
Lee Archer
- [asterisk-users] ASterisk OOH323c
Michel
- [asterisk-users] how to register nokia with Asterisk
Biju
- [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
Dan Austin
- [asterisk-users] asterisk (FreePBX) and queues
Felipe Neuwald
- [asterisk-users] SIP/TCP?
Yuan LIU
- [asterisk-users] asterisk 1.4 debian packages
Juraj Bednar
- [asterisk-users] Has anybody voipstunt working?
Arik Raffael Funke
- [asterisk-users] Multiple users and a single extension
Phil Finkler
- [asterisk-users] SIP/TCP?
James R. Stevens
- [asterisk-users] Voicemail personalised greetings using DB/IMAP
backend?
Ray Jackson
- [asterisk-users] Random "unknown" codec format IAX calls
Max Ochoa
- [asterisk-users] DiD for less then $4
CM Rahman
- [asterisk-users] Voicemail personalised greetings using
DB/IMAPbackend?
Douglas Garstang
- [asterisk-users] Asterisk to a Huawei softX3000 problem has already been solved !
Guillermo Salas M.
- [asterisk-users] Call waiting notification
Kevin Smith
- [asterisk-users] .call files no longer generating CDR files
Michael Collins
- [asterisk-users] radius
Khaled
- [asterisk-users] asterisk 1.4.0 didn't compile chan_zap.so
Martti Tienhaara
- [asterisk-users] Asterisk is used in U.S. prisons?
Ronald Lewis
- [asterisk-users] ASterisk OOH323c
Ngo Duc Loi
- [asterisk-users] Secure a Asterisk Server ?
Noc Phibee
- [asterisk-users] Handling SIP 482 condition
Chris Miller
- [asterisk-users] SIP trunk to a Boscom/Claro/IP Gear Robocom
Rob Hillis
- [asterisk-users] Hint and call-limit issue
Nick Adams
- [asterisk-users] SIP Reinvites
Adrian Marsh
- [asterisk-users] CISCO 2600 - VWIC 1MFT-E1
Andrew Pogrebennyk
- [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
Dan Austin
- [asterisk-users] Re: Best inexpensive home office router for VoIP
(QoS with maybe PoE)
Robbie Hughes
- [asterisk-users] SIP/RTP Nat problem, can't solute it.
Facundo Barrera - GMail
- [asterisk-users] Shared Line Appearances in 1.4
Marc Archer
- [asterisk-users] Parking a call a second time using #700..
Marc Archer
- [asterisk-users] Question about AGI and variable storage
Lee Jenkins
- [asterisk-users] "Reserved" extensions?
Yuan LIU
- [asterisk-users] How to get dial tone back
Yuan LIU
- [asterisk-users] Scalable IVR with asterisk
snacktime
- [asterisk-users] LUSYN patches
Marc McLaughlin
- [asterisk-users] Hanging up a 3-way conference when middle user
hangs up
Lex Lethol
- [asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 7
Naija Man
- [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released
Naija Man
- [asterisk-users] How to transfer Voicemail messages between 2
Asterisk servers
Naija Man
- [asterisk-users] Some queries on g729 license.
Xue Liangliang
- [asterisk-users] snom 360 auto answer
Jason Kim
- [asterisk-users] AMAFlags always"Documentation" (or 3 in astcdr
mysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill)
Scott Keagy
- [asterisk-users] Problems with park
Scott Walde
- [asterisk-users] Interrupt rates and voip traffic
Rajkumar S
- [asterisk-users] jitterbuffer on sip.conf
santok at student.eepis-its.edu
- [asterisk-users] Goto not jumping to current context
Dinesh Nair
- [asterisk-users] MFC/R2 problems
yusuf
- [asterisk-users] Manage 'full' log file
jan.sarin at securia.se
- SV: [asterisk-users] Manage 'full' log file
jan.sarin at securia.se
- [asterisk-users] Which is GUI to edit Asterisk IVR logic
Tzafrir Cohen
- [asterisk-users] IAX call path optimization with more than 3 legs
Ramon Schönborn
- SV: [asterisk-users] Manage 'full' log file
jan.sarin at securia.se
- [asterisk-users] Some queries on g729 license.
Matthew Rubenstein
- [asterisk-users] Block some number outgoing from joust
oneextention
Mattias Andersson
- [asterisk-users] Adding 4000 Lines to asteriskdb via asterisk -rx ?
Christoph Adomeit
- [asterisk-users] ARA extensions ordering
Jesse Peterson
- [asterisk-users] Realtime Voicemail Table Column Name Question
JR Richardson
- [asterisk-users] G729 license counting
Michel
- [asterisk-users] G729 license counting
Douglas Garstang
- [asterisk-users] Strange error
Il Neofita
- [asterisk-users] OT:spa942 provisioning
Benko
- [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released
Dan Austin
- [asterisk-users] MixMonitor write issue
Jay Moore
- [asterisk-users] Asterisk from Debian Packages
Andreas v. Heydwolff
- [asterisk-users] Strange queue behaviour
José Pablo Fernández
- [asterisk-users] Some queries on g729 license.
Darryl Dunkin
- [asterisk-users] delete=yes is not working
Mark Greene
- [asterisk-users] snom 190 (etc.?) dialscript for * debugging and
kaddressbook
Andreas v. Heydwolff
- [asterisk-users] SIP rt load from db
Tim Connolly
- [asterisk-users] Call Sound Volume Low : between extensions and
over ZAP.
Klaverstyn, David C
- [asterisk-users] Asterisk and MiniITX setups
C F
- Fwd: [asterisk-users] Some queries on g729 license.
C F