September 2006 Archives by thread
Starting: Fri Sep 1 00:03:26 MST 2006
Ending: Sat Sep 30 20:54:35 MST 2006
Messages: 3621
- [asterisk-users] asterisk presence (from manager API)
harrygaillac-sip at yahoo.fr
- [asterisk-users] quadbri & TDM400P on same pbx ?
Giorgio Incantalupo
- [asterisk-users] Any way to go from factory reset 7970 to SIP
without Call Manager?
Jason Lixfeld
- [asterisk-users] Re: Any way to go from factory reset 7970 to SIP
without Call Manager?
Tomislav Parčina
- [asterisk-users] help me!!Problem on incoming calls
Andrea infoteam
- [asterisk-users] help me!!Problem on incoming calls
Andrea infoteam
- [asterisk-users] Outgoing Call group ?
Noc Phibee
- [asterisk-users] [Slightly OT] Grandstream configurator tool
Andrea Spadaccini
- [asterisk-users] [Slightly OT] Grandstream configurator tool
Andrea Spadaccini
- [asterisk-users] Probelm with incoming calls to my DID-Please help
me
Crazy Boy
- [asterisk-users] voicemail as email and attachment
Tim St. Pierre
- [asterisk-users] 911 versus 9.911
end1r
- [asterisk-users] looking for GXV-3000 users
marek cervenka
- [asterisk-users] Sipura SPA3000
Rich Adamson
- [asterisk-users] Asterisk as a SER client
Andrea Spadaccini
- [asterisk-users] balance anouncement
ram
- [asterisk-users] Different MOH in parked calls??
equis software
- [asterisk-users] phpagi syntax and SendDTMF
Frank Church
- [asterisk-users] Re: Adit 3104 randomly reboot
Jerry Jones
- [asterisk-users] vim syntax highlighting( for Asterisk.conf files)
Marco Mouta
- [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes
Kevin Smith
- [asterisk-users] Missing Agent Function
William Piper
- [asterisk-users] question of CLI
William Piper
- [asterisk-users] Asterisk Core dump
Anthony Musaluke
- [asterisk-users] 911 versus 9.911
Jay Milk
- [asterisk-users] Toll-Free numbers
Jay Milk
- [asterisk-users] incompatible hardware?
Roy Sigurd Karlsbakk
- [asterisk-users] US Toll-Free DID Providers with Caller ID NAME?
Jay Milk
- [asterisk-users] Sipura 3000 and Asterisk
Francisco Seratti
- [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060
Elpidio Ramos
- [asterisk-users] Hardware ? Analog DID trunks (ILT)
Jonn R Taylor
- [asterisk-users] SPA-942 Sound Quality
Andres
- [asterisk-users] Asterisk speaks Italian!
Stuart
- [asterisk-users] Callback + dtmf problem
Insider KT
- [asterisk-users] Hardware ? Analog DID trunks (ILT)
Tim Sharp
- [asterisk-users] Cisco 7960 won't download dialplan.xml
Peter Pauly
- [asterisk-users] Sipura SPA3000
Rich Adamson
- [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware
Richard Klingler
- [asterisk-users] Operator Console(s)/Shared Call Appearances
Mr. Jones
- [asterisk-users] Can QUEUE member be assigned from a GlobalVar set
in EXTENSIONS.CONF?
Gary G. Hendershot
- [asterisk-users] Fax with asterisk?
Matthias Fechner
- [asterisk-users] What does 'trunk' mean in outgoing and incoming?
Larry Alkoff
- [asterisk-users] Asterisk & Linksys PAP2 ATA
Nick Ellson
- [asterisk-users] Help with blind transfer
George A. Roberts IV
- [asterisk-users] Unable to open pseudo channel for timing... Sound
may be choppy.
Dominik Weber
- [asterisk-users] Blind transfer 3/4 digits
Ronald Wiplinger
- [asterisk-users] "best" BRI card ?
Tzafrir Cohen
- [asterisk-users] Asterisk mysql cdr
Abdul
- [asterisk-users] G729 Replacement Codec - FREE or may ne cheaper
than existing one.
Kannaiyan Natesan
- [asterisk-users] Keys pressed not registering ...
Lenny
- [asterisk-users] Help compiling asterisk-addons on Debian?
Rushowr
- [asterisk-users] Re: [asterisk-dev] G729 Replacement Codec - FREE
or may ne cheaper than existing one.
Kannaiyan Natesan
- [asterisk-users] Re: Cisco 7960G SIP firmware 8.4
Nathan Alberti
- [asterisk-users] Re: [asterisk-dev] G729 Replacement Codec - FREE
or may ne cheaperthan existing one.
Kannaiyan Natesan
- [asterisk-users] Help compiling asterisk-addons on Debian?
Tzafrir Cohen
- [asterisk-users] Asterisk server crashes after two years
Tzafrir Cohen
- [asterisk-users] Problems compil 1.2.11
Tzafrir Cohen
- [asterisk-users] Nokia N80
Dean Collins
- [asterisk-users] Problems compil 1.2.11
Andrea Spadaccini
- [asterisk-users] Keys pressed not registering ...
Lenny
- [asterisk-users] Re: [asterisk-biz] G729 Replacement Codec - FREE
or may ne cheaper than existing one.
Kannaiyan Natesan
- [asterisk-users] Re: [asterisk-biz] G729 Replacement Codec - FREE
or may ne cheaper than existing one.
Kannaiyan Natesan
- [asterisk-users] [asterisk-dev] Re: [asterisk-biz] G729 Replacement
Codec - FREE ormay ne cheaper than existing one.
Kannaiyan Natesan
- [asterisk-users] res_osp.c not compiled
neil
- [asterisk-users] Caller ID has extra digits to strip
Bart Fisher
- [asterisk-users] How to send correct Caller ID on PRI
Zeeshan Zakaria
- [asterisk-users] SIPP problem
Diego Quintana Cruz
- [asterisk-users] Roundrobin not working on PRI
Zeeshan Zakaria
- [asterisk-users] How to use Grandstream GX-2000 phones for paging
Zeeshan Zakaria
- [asterisk-users] Queue timeout problems
Mr. Jones
- [asterisk-users] Queue timeout problems
Guido Hecken
- [asterisk-users] What I always get asked in SME * deployments
Eric Bishop
- [asterisk-users] SER+Asterisk integration
Siqhamo Sifo
- [asterisk-users] File structure question
Tzafrir Cohen
- [asterisk-users] Queue timeout problems
Guido Hecken
- [asterisk-users] Asterisk not sending RTP
Nir Simionovich
- [asterisk-users] Using Thunderbird (mail client) to call Contacts
from Address Book
Matt Riddell (IT)
- [asterisk-users] UK (BT) Problem with TDM 400P
Mark Muffett
- [asterisk-users] Asterisk+ser+docs
Siqhamo Sifo
- [asterisk-users] Query about Call Detail Record in Asterisk
Chan Kwang Mien
- [asterisk-users] AgentCallBackLogin and cdrupdate
Garth van Sittert
- [asterisk-users] SIPP problem
Diego Quintana Cruz
- [asterisk-users] PBX -> VoIP migration
Richard Klingler
- [asterisk-users] Please help route incoming PSTN calls to Asterisk
Larry Alkoff
- [asterisk-users] Asterisk calling through FWD?
Nick Ellson
- [asterisk-users] Re:sip giving problems, please help.
Ma Zhiyong
- [asterisk-users] Fax with asterisk?
DRi at b-w-computer.de
- [asterisk-users] No more linux/compiler.h in Fedora Core 6.
William F. Acker WB2FLW +1-303-722-7209
- [asterisk-users] Prompts playback changing tempo w/ SMP kernel
RR
- [asterisk-users] "Asterisk Developers Mailing List"
<asterisk-dev@lists.digium.com>,
Vidura Senadeera
- [asterisk-users] Zaptel-1.2.8 compile problem
Vidura Senadeera
- [asterisk-users]
Architecture:MainOffice(AstServer)-SmallOffices(ATA.-LegacyPBX)
Marco Mouta
- [asterisk-users] Any Hardphone with VPNClient embedded?
Marco Mouta
- [asterisk-users] External calls from Asterisk over a
Siemens(legacy) RDSI PBX
Llorenç Suau
- [Asterisk-Users] External calls from Asteris over a legacy Siemens
BusinessPhone 250 PBX
Llorenç Suau
- [asterisk-users] includes in realtime ??
Benjamin Jacob
- [asterisk-users] usereqphone=yes seems to don't work
Alexandre VERNIOL
- [asterisk-users] Asterisk 1.2.11 and # key
Michael Strelnikov
- [asterisk-users] FAX handling
Jose Limeres
- [asterisk-users] PAP2-NA + Asterisk
joy panlilio
- [asterisk-users] Handling Disconnection Causes
Rafael J. Risco G.V.
- [asterisk-users] Submenus
Mir
- [asterisk-users] Dropping extra frame of G.729 ?
Noc Phibee
- [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK
Elpidio Ramos
- [asterisk-users] missing pri connect (wwomera to pri)
Rosario Pingaro
- [asterisk-users] blf aastra 9133i working but can't pickup calls
Jean-Louis curty
- [asterisk-users] Astbill DIALSTRING doesn't work
Sebastian Milioto
- [asterisk-users] playback some digits to the caller from the callee
(involves DTMF) prob
umar tarar
- [asterisk-users] Grandstream and H.264 !
Sergio (Red)
- [asterisk-users] Looks like Nufone is changing around...
Justin Newman
- [asterisk-users] app_conference not working for me
Steve Edwards
- [asterisk-users] Call center reports
Technical Support
- [asterisk-users] Re: Nufone making changes
Justin Newman
- [asterisk-users] Asterisk calling through FWD?
Nick Ellson
- [asterisk-users] Blind transfer 3/4 digits
wendell hamilton
- [asterisk-users] Digum g729 and g723
Joe shmoe
- [asterisk-users] SNMP with 1.2.11 stable
Azfhasterisk
- [asterisk-users] Asterisk calling through FWD?
Nick Ellson
- [asterisk-users] Re: FAX handling
Justin Newman
- [asterisk-users] Warning about using PAP2-NA ATA recent firmware
3.1.12 LS
joy panlilio
- [asterisk-users] HITBSecConf2006 Final Call !
Praburaajan
- [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware
Tomislav Parčina
- [asterisk-users] Reading the raw E1 channels ?
Azher Amin
- [asterisk-users] Re: [asterisk-biz] Re: G729 Replacement Codec -
FREE or may ne cheaper than existing one.
Kannaiyan Natesan
- [asterisk-users] End of call
Michael Strelnikov
- [asterisk-users] connect with two servers multiple time
Arjan Kroon
- [asterisk-dev] Re: [asterisk-users] Digum g729 and g723
Kannaiyan Natesan
- [asterisk-users] Re: How to use Grandstream GX-2000 phones for
paging
Tomislav Parčina
- [asterisk-dev] Re: [asterisk-users] Digum g729 and g723
Kannaiyan Natesan
- [asterisk-users] A couple more interviews with Digium staff
Matt Riddell (IT)
- [asterisk-users] why executed Hangup doesn't exit DialPlan?look my
dialplan...
Marco Mouta
- [asterisk-users] Can not hear the telco System Announcement
stoffell
- [asterisk-users] How to manipulate a plus in a phone number
Tim Panton
- [asterisk-users] latest CentOS-asterisk-freepbx installation
procedure
Roland
- [asterisk-users] Experience Patton BRI gateways and Asterisk?
Koopmann, Jan-Peter
- [asterisk-users] telco error message on PRI and BRI
stoffell
- [asterisk-users] LinkSys PAP2 ATA & Siemens Cordless 3010
Jopi at fastwebnet.it
- [asterisk-users] ISDN config EWSD
Virmones Pereira Tavares de Miranda
- [asterisk-users] Zero length queue
Artifex Maximus
- R: Re: [asterisk-users] LinkSys PAP2 ATA & Siemens Cordless 3010
Jopi at fastwebnet.it
- [asterisk-users] Keys pressed not registering ...
Lenny
- [asterisk-users] includes in realtime ??
Douglas Garstang
- [asterisk-users] Unable to make calls from CallManager to Asterisk
Anantha Padmanabha.M.L
- [asterisk-users] Different MOH in waiting calls and parked calls
equis software
- [asterisk-users] Experience Patton BRI gateways and Asterisk?
Guido Hecken
- [asterisk-users] monoBRI + install-misdn-mqueue: no inbound calls
but strange messages
Giorgio Incantalupo
- [asterisk-users] ATA being used as a SIP Trunk to connect LegacyPbx
to Main Asterisk Server
Marco Mouta
- [Asterisk-Users] T1 echo canceller
Michael Araba
- [asterisk-users] Find-Me/Follow-ME
Roger Workman
- [asterisk-users] Asterisk vicidial question
Gustavo Alejandro Gonzalez
- [asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes
Timothy R. McKee
- [asterisk-users] Asterisk vicidial question
ggonzalez at telviso.com.ar
- [asterisk-users] Find-Me/Follow-ME
Joel Vandal
- [asterisk-users] Faxing ..
Lenny
- [asterisk-users] Find-Me/Follow-ME
Roger Workman
- [asterisk-users] Different MOH between waiting calls and transfer
calls
equis software
- [asterisk-users] IAX and rsa
andrutto
- [asterisk-users] Meet-me recording formats
Michael Lively
- [asterisk-users] config include issues
Curt Shaffer
- [asterisk-users] asterisk t.38 fax failed
Kokfoo Soo
- [asterisk-users] Linking Asterisk with PBX through E1
Marlon Dutra
- [asterisk-users] Adding custom fields (more than one) to CDR DB
Mike
- [asterisk-users] Is this a warning or not...MYSQL Fetch
Mike
- [Asterisk-Users] Asterisk Cygwin Port.
joshnet at nbnet.nb.ca
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- [asterisk-users] How to notify an ACD agent before he/she picks up
Manrique Feoli
- [asterisk-users] Deadlock
Michael Welter
- [asterisk-users] Merlin Legend - Working Now!
Sterling Moses
- [asterisk-users] Native Chinese speaker needed
John Williams
- [asterisk-users] Need somebody for video phone testing
Ronald Wiplinger
- [asterisk-users] Has anyone tried to install both digital card and
analog card in one machine
Xue Liangliang
- [asterisk-users] Really bad phone line.. possible causes?
Jeff Turner
- [asterisk-users] macros in Realtime
Benjamin Jacob
- [Asterisk-Users] T1 echo canceller
Michael Araba
- [asterisk-users] Asterisk + Samsung OffServ 500
Eugeniy Khvastunov
- [asterisk-users] Dell Poweredge SC430 and Digium cards
compatability enquiry
Matthew Thompson
- [asterisk-users] core dumps
Anthony Musaluke
- [asterisk-users] Asterisk AGI and Firebird
Steve Rawlings
- [asterisk-users] How to test TE405P T1
Andy Chung (Power-All)
- [asterisk-users] Budgetones - multiple phones losing IP address
during day
Garth van Sittert
- [asterisk-users] sangoma A104d echo canceller and fax
Klaus Darilion
- [asterisk-users] flag 'g' in Dail() is'nt working with
agentcallbacklogin()
umar tarar
- [asterisk-users] mobile refusing call
René Enskat [Teamware GmbH]
- [asterisk-users] how to setup poxy sip server
Ranjeet Kumar
- [asterisk-users] Answer Machine detection
Mark Ackroyd
- [asterisk-users] how to setup poxy sip server
brian at neotiq.com
- [asterisk-users] cmd SET time value
Benjamin Jacob
- [asterisk-users] SIP client with video???
Joao Pereira
- [asterisk-users] Asterisk video support
Joao Pereira
- [asterisk-users] app_rxfax Only Receives One Page
Steve Totaro
- [Asterisk-Users] Which SIP hardphone with embedded VPNClient ?
Olivier
- [asterisk-users] Cisco PIX firewall and nat=yes
Bill Gibbs
- [asterisk-users] Cisco MWI
Steve Kennedy
- [asterisk-users] Cisco PIX firewall and nat=yes
Bill Gibbs
- [asterisk-users] Conditional IF based on IP address?
Steve Hsieh
- [asterisk-users] Is asterisk's mgcp support(NAS) Network access
server package
Ibrar Ahmed
- [asterisk-users] Is asterisk's mgcp support(NAS) Network
access server package
Davor Grgicevic
- [asterisk-users] Call parking and RTP traffic
Dave Fullerton
- [asterisk-users] Cisco MWI
Doug Lytle
- [asterisk-users] Linksys SPA-942 + Asterisk 1.2.10 = Inability to
transfer calls
Dan Serban
- [asterisk-users] Volume events causing talk off on Asterisk with
Digium 411P
Servetas, Andrew
- [asterisk-users] faktortel
Dean Collins
- [asterisk-users] Digium G.729 codec binaries updated
Kevin P. Fleming
- [asterisk-users] Digium's response to posting of G.729 and G.723
source code
Kevin P. Fleming
- [asterisk-users] Garbled (quality probs) IAX2 & SIP calls
Asterisk-to-Asterisk
lists.digium.com at tgice.com
- [asterisk-users] using SIP to connect remote other VoIP server
tengulre
- [asterisk-users] Polycom new firmware and bootrom
Chris Dos
- [asterisk-users] How to check which rtp ports my firewall let
through?
Ronald Wiplinger
- [asterisk-users] the sounds quality of IAX2 channels are not good
as SIP channels?
Ma Zhiyong
- [asterisk-users] cmd SET time value
Benjamin Jacob
- [asterisk-users] Re: [asterisk-dev] UUI in calls
John Todd
- [asterisk-users] Response to KP Flemming...
Joe Shmoe
- [asterisk-users] ast_parse_allow_disallow: Cannot allow unknown
format 'h264'
Ronald Wiplinger
- [asterisk-users] Response to KP Flemming...
Joe Shmoe
- [asterisk-users] Response to KP Flemming...
Joe Shmoe
- [asterisk-users] How to send and receiving fax with asterisk?
Andrea infoteam
- [asterisk-users] Configuring new IAX2 Jitter Buffer for IVR
application.
John Melody
- [asterisk-users] New polycom firmware / presence
harrygaillac-sip at yahoo.fr
- [asterisk-users] netmask
Dean Collins
- [asterisk-users] WG: mobile refusing call
René Enskat [Teamware GmbH]
- [asterisk-users] bristuff compile problems with kernel 2.6.17.11
Arik Raffael Funke
- [asterisk-users] Softphones IAX vs. SIP, remote connectivity.
Nick Ellson
- [asterisk-users] Cisco 7970 directories and services xml
Tomislav Parčina
- [asterisk-users] Incoming call problem-calling part is busy(IPKall)
Crazy Boy
- [asterisk-users] Re: Cisco 7970 8.0.4 SIP firmware
Tomislav Parčina
- [asterisk-users] Experiences,
Tips on Voicemail storage using ODBC or IMAP?
Rushowr
- [asterisk-users] Asterisk "Clusters"
Mitch Thompson
- [asterisk-users] Incoming call problem-calling part is busy(I
PKall)
Guido Hecken
- [asterisk-users] RE: Volume events causing talk off on Asterisk
with Digium 411P
Servetas, Andrew
- [asterisk-users] Re: Cisco 7970 directories and services xml
Tomislav Parčina
- [asterisk-users] Re: Cisco 7970 directories and services xml
Tomislav Parčina
- [asterisk-users] Capacity for transcode G711 to G729
Kokfoo Soo
- [asterisk-users] New polycom firmware / presence
Douglas Garstang
- [asterisk-users] svn trunk or branches ???
Ronald Wiplinger
- [asterisk-users] g729 failover when out of licenses
Tod Detre (CampusEAI Consortium)
- [asterisk-users] Voicemail Delete Bug?
Douglas Garstang
- [asterisk-users] blf aastra 9133i working but can't pickup calls
Gareth Owen
- [asterisk-users] Asterisk and NAT ?
Noc Phibee
- [asterisk-users] Asterisk hangs up after 10-15 minutes when SIP
Phone is on mute
Mike
- [asterisk-users] Softphones IAX vs. SIP, remote connectivity.
Ferguson, Michael
- [asterisk-users] uConnect Voip device
Frank Church
- [asterisk-users] Sound (or lack of it) problems
Jordan Kirby
- [asterisk-users] 0005162: RTP Packetization : Few questions
yusuf
- [asterisk-users] Polycom new firmware and bootrom
Douglas Garstang
- [asterisk-users] How to Install H323
Wasif
- [asterisk-users] 0005162: RTP Packetization : Few questions
Dan Austin
- [asterisk-users] Softphones IAX vs. SIP, remote connectivity.
Ferguson, Michael
- [asterisk-users] Softphones IAX vs. SIP, remote connectivity.
Ferguson, Michael
- [asterisk-users] Asterisk Outgoing Spool Failed
Arun Kumar
- [asterisk-users] Polycom new firmware and bootrom
Douglas Garstang
- [asterisk-users] Re: Volume events causing talk off on Asterisk
with Digium 411P
Servetas, Andrew
- [asterisk-users] Softphones IAX vs. SIP, remote connectivity.
Ferguson, Michael
- [asterisk-users] 0005162: RTP Packetization : Few questions
Dan Austin
- [asterisk-users] Polycom new firmware and bootrom
Douglas Garstang
- [asterisk-users] Softphones IAX vs. SIP, remote connectivity.
Ferguson, Michael
- [asterisk-users] TDM400 and T100 config on same asterisk
Rich
- [asterisk-users] Open source G.729 and G.723.1 release for 1.2 and
1.4
Daniel Pocock
- [asterisk-users] Speex Codex - Eyebean to Asterisk
Kokfoo Soo
- [asterisk-users] Call Forwarding in SIP.conf
broadbandvoice at comcast.net
- [asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 39
Servetas, Andrew
- [asterisk-users] RE: Volume events causing talk off on Asterisk
with Digium 411P
Servetas, Andrew
- [asterisk-users] te110p and te205p behavioural differences
Mark Edwards
- [asterisk-users] Intel 945G and Digium TE110P compatibility issue
Xue Liangliang
- [asterisk-users] Asterisk 1.2 and SATA drives
Tharanga
- [asterisk-users] app_amd and voicemail
David Koski
- [asterisk-users] Call Forwarding in SIP.conf
broadbandvoice at comcast.net
- [asterisk-users] dialplan applications
Robert Bielik
- [asterisk-users] Asterisk 1.2 and SATA drives
Guido Hecken
- [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990
harrygaillac-sip at yahoo.fr
- [asterisk-users] New polycom firmware / presence
harrygaillac-sip at yahoo.fr
- [asterisk-users] sip peer question
Dijkstra, Roelof
- [asterisk-users] Problems with app_directed_pickup
Jon Schøpzinsky
- [asterisk-users] Digits are played in english in french voicemail
harrygaillac-sip at yahoo.fr
- [asterisk-users] Trouble with rxfax multi-page printing with cups
Artifex Maximus
- [asterisk-users] Digits are played in english in french voicemail
Dijkstra, Roelof
- [asterisk-users] Asterisk and "Maximum retries exceeded"
Noc Phibee
- [asterisk-users] distinguishing users by their domain
Ricardo Carvalho
- [asterisk-users] Problems with KG1000 voip gateway and DTMF
Terence Haddock
- [asterisk-users] Grandstream GX-2000,
doesn't send calls to free lines
Zeeshan Zakaria
- [asterisk-users] How can I set CDR data in dialplan?
Set(CDR(src)=foo)
Mike
- [asterisk-users] Grandstream, how to use the configuration tool
Zeeshan Zakaria
- [asterisk-users] Tracking the source of a disconnect?
Jamin W. Collins
- [asterisk-users] FW: Peter Dicks Chairman of Sportingbet PLC is
arrested at JFK!!
Dean Collins
- [asterisk-users] Reload question
Mailing List
- [asterisk-users] Asterisk and SIP Redirect message
Michel Zenone
- [asterisk-users] FW: Peter Dicks Chairman of Sportingbet PLC is
arrested at JFK!!
Dean Collins
- [asterisk-users] Call Forwarding in SIP.conf
broadbandvoice at comcast.net
- [asterisk-users] 0005162: RTP Packetization : Few questions
Dan Austin
- [asterisk-users] Use PauseQueueMember
gc
- [asterisk-users] Want to support a better SIP stack in Asterisk?
Olle E Johansson
- [asterisk-users] ISDN HFC card cannot 'detect remote answer'
Edoardo Serra
- [asterisk-users] Call Forwarding in SIP.conf
broadbandvoice at comcast.net
- [asterisk-users] What don't I get about SIP?
Mike
- [asterisk-users] No dialtone, just directly busy
Henrik Woffinden
- [asterisk-users] RE: Peter Dicks Chairman ofSportingbet PLC is
arrested at JFK!!
Dean Collins
- [asterisk-users] help chan_bluetooth
Mauricio Mantilla
- [asterisk-users] I'm I wrong - No 3-way calling for Single line
sets?
Bart Fisher
- [asterisk-users] Little help for a newbie configuring a TDM13B -
ztcfg fails on channel 4
Iván Vega R.
- [asterisk-users] Re: Little help for a newbie configuring a TDM13B
- ztcfg fails on channel 4
Iván Vega R.
- [asterisk-users] No such device -> TDM13B
Iván Vega R.
- [asterisk-users] Stupid question about FXS/FXO
Iván Vega R.
- [asterisk-users] MSSQL connection
Kevin Smith
- [asterisk-users] How to play a sound on a periodic basis during a
call?
Dallas Carter
- [asterisk-users] Re: No such device -> TDM13B
Iván Vega R.
- [asterisk-users] Asterisk 1.2.12 and Zaptel 1.2.9 released!
The Asterisk Development Team
- [asterisk-users] zaptel 1.2.9 won't compile
Earl Terwilliger
- [asterisk-users] Zaptel-1.2.9 compile error
Bill Maidment
- [asterisk-users] Receive Fax with rxfax on asterisk with debian
Dominik Weber
- [asterisk-users] Call Forward Problem
James Williams
- [asterisk-users] Call forwarding
Vladimir Dvorak
- [asterisk-users] Call Forwarding in SIP.conf
broadbandvoice at comcast.net
- [asterisk-users] Call Processing Slow 11 seconds
broadbandvoice at comcast.net
- [asterisk-users] Another (quick) Polycom 501 question
Mike
- [asterisk-users] ISDN / Multiplink PPP (ZapRAS)
Brent Franks
- [asterisk-users] DID Provider in Thailand
broadbandvoice at comcast.net
- [asterisk-users] Call Forwarding in SIP.conf
broadbandvoice at comcast.net
- [asterisk-users] Intel Based G.729 and SVN-trunk-r42453
Jason Lee
- [asterisk-users] Call Forwarding in SIP.conf
broadbandvoice at comcast.net
- [asterisk-users] Call Processing Slow 11 seconds
broadbandvoice at comcast.net
- [asterisk-users] RE: asterisk-users Digest, Vol 26, Issue 54
FRANCISCO PEREZ-LANDAETA
- [asterisk-users] Call Processing Slow 11 seconds
broadbandvoice at comcast.net
- [asterisk-users] Problems configuring Polycom 301
Jim Freeze
- [asterisk-users] ztdummy installed but choppy audio warning on load
Nigel Godfrey
- [asterisk-users] Using option 'r' in queue doesn't announce
frequeny etc.
Zeeshan Zakaria
- [asterisk-users] Whcih phones are better for mass deployment
Zeeshan Zakaria
- [asterisk-users] Scope of contexts
Rene
- [asterisk-users] Whcih phones are better for mass deployment
broadbandvoice at comcast.net
- [asterisk-users] What really happens between Asterisk and an
SPA-3000?
Larry Alkoff
- [asterisk-users] Whcih phones are better for mass deployment
broadbandvoice at comcast.net
- [asterisk-users] Grandstream GX-2000 Remote Login Problem
Zeeshan Zakaria
- [asterisk-users] Streaming audio for MoH
Zeeshan Zakaria
- [asterisk-users] Quintum tenor configuration with asterisk help
FRANCISCO PEREZ-LANDAETA
- [asterisk-users] call notification for queues?
Rajkumar S
- [asterisk-users] Satellite link-IAX Jitter Buffer.
ANANGWE Nelson
- [asterisk-users] How could i get bridged channel partner
Mohammad Shokuie
- [asterisk-users] Quintum tenor configuration with asterisk help
c.savinovich at itntelecom.com
- [asterisk-users] Take 3 -- Trying to get SIP firmware on a 7970G
Jason Lixfeld
- [asterisk-users] Polycom related question
Kevin Smith
- [asterisk-users] Accounts registered, but call is not going
Crazy Boy
- [asterisk-users] su - postgres -bash-3.00$
broadbandvoice at comcast.net
- [asterisk-users] QUINTUM CONFIGURATION.-
FRANCISCO PEREZ-LANDAETA
- [asterisk-users] Voip providers and sip origination and termination?
Christopher Corn
- [asterisk-users] can someone recommend a voip provider that...
Christopher Corn
- [asterisk-users] music onhold choppy music problems
Matt
- [asterisk-users] can someone recommend a voip provider that...
broadbandvoice at comcast.net
- [asterisk-users] data pass through
robb at boardman.me.uk
- [asterisk-users] Max Size of Conf Files
Steve Totaro
- [asterisk-users] using residential voip for business?
Christopher Corn
- [asterisk-users] Setting system time via Asterisk
Gary Eck
- [asterisk-users] using residential voip for business?
broadbandvoice at comcast.net
- [asterisk-users] Hrmm.. OK,
what am I missing? sendmail: Cannot open mail:25
Nick Ellson
- [asterisk-users] QUINTUM TENOR ASM200 Configuration
FRANCISCO PEREZ-LANDAETA
- [asterisk-users] Sound Quality.
Joel Hill
- [asterisk-users] Looking to hire somebody to setup a SER load
balancer
Deon
- [Asterisk-Users] Asterisk Native Sound Distortion (ulaw)
Tzafrir Cohen
- [asterisk-users] beginners question....
Panagiotis Zikos
- [asterisk-users] Context
Khaled Chehab
- [asterisk-users] How to integrate freepbx with a2billing?
Sharon Lim
- [asterisk-users] Asterisk Realtime Arch - static or realtime?
Benjamin Jacob
- [Asterisk-Users] SIP parameter to prevent a call from being added in
missed calls logs
Olivier
- [asterisk-users] Problems with outgoing calls
Roy Gardner
- [asterisk-users] Outgoing callerid in AMI
Mir
- [Asterisk-Users] SIP hardphones and BLF monitoring keys
Olivier
- [asterisk-users] TE411P or TE412P?
Tony Mountifield
- [asterisk-users] I am not getting 302 redirects...
Arik Raffael Funke
- [asterisk-users] modifying the INVITE headers
Paco Brufal
- [asterisk-users] Can Asterisk bind on multiple ports?
Ricardo Carvalho
- [asterisk-users] Ringtones
Scott Pinhorne
- [asterisk-users] SIP trunk
Richard Klingler
- [asterisk-users] Handling incoming calls from VoIPbuster
Marco Mouta
- [asterisk-users] [Serusers] MS LCS 2005 / SER / Asterisk Integration
harrygaillac-sip at yahoo.fr
- [asterisk-users] realtime static config include contexts
Benjamin Jacob
- [asterisk-users] Quintum tenor configuration with asterisk help
Mohammad Shokuie
- [asterisk-users] su - postgres -bash-3.00$
broadbandvoice at comcast.net
- [asterisk-users] su - postgres -bash-3.00$
broadbandvoice at comcast.net
- [asterisk-users] switching from IAX to SIP
Jerry Geis
- [asterisk-users] Problems Unpacking tarball For Asterisk Application
broadbandvoice at comcast.net
- [asterisk-users] Register 2 times with same host
Richard Klingler
- [asterisk-users] Problems Unpacking tarball For Asterisk
Application
broadbandvoice at comcast.net
- [asterisk-users] Problems Unpacking tarball For Asterisk
Application
broadbandvoice at comcast.net
- [asterisk-users] Support for Intel Boards On Asterisk
cleviton.araujo at caixa.gov.br
- [asterisk-users] Verify Database Installation
broadbandvoice at comcast.net
- [asterisk-users] PRI: sometimes Asterisk drop calls
Giorgio Incantalupo
- [asterisk-users] help connecting cell phone, chan_bluetooth
Mauricio Mantilla
- [asterisk-users] Is anybody using autofill option in queue.conf?
equis software
- [asterisk-users] Grandstream Budgetone phones don't show
alphanumeric caller right
Ricardo Carvalho
- [asterisk-users] --- Dlink DVC-2000 VideoPhone (H.323) with
Asterisk ---
Ed Greenberg
- [asterisk-users] Realtime Queues and Postgres.
Fernando Lujan
- [asterisk-users] Getting Incoming called from trxtel.com
Howard Leadmon
- [asterisk-users] sip and iax over the internet (asterisk to
asterisk) drop outs normal???
Jerry Geis
- [asterisk-users] updated zaptel tarball
Steve Kennedy
- [asterisk-users] Remote tone access
Tony Di Bona
- [asterisk-users] PRI channel hangup
Michael Welter
- [asterisk-users] How to configure Fritz ISDN2 card with Trixbox?
Crazy Boy
- [asterisk-users] IAX2 trunk problem
Enrico Pasqualotto
- [asterisk-users] Polycom Soundpoint Key Remap
Shawn Kelley
- [asterisk-users] Weird (bri)stuff 0.3.0-PRE-1s
Remco Barendse
- [asterisk-users] Cable Systems ICS-G302,
Anyone have an Admin Guide Please?
JR Richardson
- [asterisk-users] Receive Fax with rxfax on asterisk with debian
Bill Gibbs
- [asterisk-users] More Zaptel build problems
Kristian Kielhofner
- [asterisk-users] experience with axvoice.com?
Christopher Corn
- [asterisk-users] Static RealTime - SIP.CONF
Hugo
- [asterisk-users] SIP DOMAIN SUPPORT
harrygaillac-sip at yahoo.fr
- [asterisk-users] g729 problem
o o
- [asterisk-users] Forward recorded voicemail message to more than
one extension using sendvoicemail=yes
Jay Dutt
- [asterisk-users] Dell hardware ...
Alan Bunch
- [asterisk-users] question...
Christopher Corn
- [asterisk-users] BLF via metermaid on 1.2.7.1 and aastra 9133i
mike pham
- [asterisk-users] SIP 415 messagse
Mr. Jones
- [asterisk-users] Change Payload
Yelson Vivas
- [asterisk-users] Bad number - is not in inbound speed dial
Enrico Pasqualotto
- [asterisk-users] How to notify an ACD agent before he/she picks up
Watkins, Bradley
- [asterisk-users] GXP2000 - Blind Transfer Hangs Up Call
Daniel Salama
- [asterisk-users] Digium at Ohio Linuxfest
Greg Boehnlein
- [asterisk-users] Polycom HD Voice - 16 Khz - Asterisk support ?
David Gagnon
- [asterisk-users] DID not getting passed?
Christopher Corn
- [asterisk-users] Polycom Soundpoint Key Remap
Douglas Garstang
- [Asterisk-Users] Sirrix BRI errors
Klaus Darilion
- [asterisk-users] Re: MSSQL connection
Tomislav Parčina
- [Asterisk-Users] Browsing distant missed call list
Olivier
- [Asterisk-Users] The best way to design local-only off-hours ringing
Olivier
- [asterisk-users] Re: Take 3 -- Trying to get SIP firmware on a 7970G
Tomislav Parčina
- [asterisk-users] about 'zap show channels'
PSPunch
- [asterisk-users] asterisk logging per day
Christophorus Laube
- [asterisk-users] SIP/2.0 403 Relaying denied
Rene
- [asterisk-users] Samsung OfficeServ 500 + Asterisk(Tormenta 2) via
PRI
Eugeniy Khvastunov
- [Asterisk-Users] Junghanns BRI cards and misdn
Olivier
- [asterisk-users] Verify Database Installation
broadbandvoice at comcast.net
- [Asterisk-Users] Which SIP hardphone implements RTCP XR (aka RFC3611)
Olivier
- [asterisk-users] Rack for Asterisk with TDM2400 Digium board
Antoine Megalla
- [asterisk-users] WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring
requested on channel
0/1 already in use on span 1. Hanging up owner.
Giorgio Incantalupo
- [asterisk-users] Features.. phone vs. asterisk?
Nick Ellson
- [asterisk-users] WG: Asterisk and Agents
mbodbg at gmx.net
- [asterisk-users] How to setup announce attibute in queues.conf
gc
- [asterisk-users] Deploying an IVR - direct extensions.conf or AGI
scripts?
Marco Mouta
- [BULK] Re: [asterisk-users] Prompts recording for Asterisk
Savoy, Kevin - Williston, ND
- [asterisk-users] Polycom Soundpoint Key Remap
Douglas Garstang
- [asterisk-users] RE : Re: [asterisk-dev] Forwarding sip requests
from none local domains
harrygaillac-sip at yahoo.fr
- [asterisk-users] Grandstream Budgetone phones don't show
Jessee Holmes
- [Asterisk-Users] Suggestion for directed pickup in bristuffed 1.2
Asterisk
Olivier
- [asterisk-users] RE : Re: [asterisk-dev] Forwarding sip requests
from none localdomains
harrygaillac-sip at yahoo.fr
- [asterisk-users] IAX phone recommandation
richard Coco
- [asterisk-users] Dropped call question - "Maximum retries exceeded
on transmission"
Kohler, Jeffrey
- [asterisk-users] Conference bridge problem
Bartosz Wegrzyn - asterisk
- [asterisk-users] Problems getting 7970G upgraded to SIP
Jason Lixfeld
- [asterisk-users] Polycom MyStat
Douglas Garstang
- [asterisk-users] Verizon ISDN service in NY & Hunt Groups
Bernie Courtney
- [asterisk-users] Calling Card and Billing
sdcharly at gmail.com
- [asterisk-users] A simple goal, help me please!
David R.
- [asterisk-users] test
harrygaillac-sip at yahoo.fr
- [asterisk-users] AEL if/else/IFTIME fun.
Dan Serban
- [asterisk-users] Trouble connecting to my telco with fonebridge
Leif Kristian Hetlesæther
- [asterisk-users] RE: [asterisk-biz] Come see us at VON
Dean Collins
- [asterisk-users] Polycom Soundpoint Key Remap
Shawn Kelley
- [asterisk-users] Please help with a telular mod. SX5e
Jorge Cisneros
- [asterisk-users] consitent half channel loss after 6 minutes
Jerry Geis
- [asterisk-users] strange problem with calls between MGCP and SIP
clients(ATA's)
Andy Kuo
- [asterisk-users] Polycom Soundpoint Key Remap
Douglas Garstang
- [asterisk-users] Switch Experiences
Ben Gore
- [asterisk-users] sip origination and termination
Christopher Corn
- [asterisk-users] Polycom Soundpoint Key Remap
Watkins, Bradley
- [asterisk-users] sip origination and termination
broadbandvoice at comcast.net
- [asterisk-users] Re: [A*UG] What does 'trunk' mean in outgoing and
incoming?
Larry Alkoff
- [asterisk-users] All circuits are busy now???
BerkHolz, Steven
- [asterisk-users] sound file length
Raphael Jacquot
- [asterisk-users] INX (internationalnumber.com) Outgoing problem
Daniel Cyt
- [asterisk-users] Virtualise asterisk on Xen
Arik Raffael Funke
- [asterisk-users] Makefile.moddir_rules: No such file or directory
Ronald Wiplinger
- [asterisk-users] Polycom Firmware
Forum Expansive
- [asterisk-users] fxotune failure! "Could not fill input buffer -
got -1 bytes, expected 4000 bytes"
Jonathan Barratt
- [asterisk-users] call waiting
Christopher Corn
- [asterisk-users] Huh? An ilbc frame that isn't a multiple of 50 bytes
long from RTP (4)?
Andy Kuo
- [asterisk-users] Re: Features.. phone vs. asterisk?
Tomislav Parčina
- [asterisk-users] Long Delay in IAX Calls
Jonathan Palley
- [asterisk-users]
Anyone working on VXML, CCXML support for asterisk?
Arnd Vehling
- [asterisk-users] Queue - persistent members
Tomislav Parčina
- [asterisk-users] Polycom IP430 sound level too low?
Louis-David Mitterrand
- [asterisk-users] "Too many files..." error - best way to fix?
Matt Arnilo S. Baluyos (Mailing Lists)
- [asterisk-users] Kirk IP600 V3 DECT Wireless server
Remco Barendse
- [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya
Definity Experts out there?)
Steve Totaro
- [asterisk-users] Choppy MOH (Cisco gateway)
Zeeshan Zakaria
- [asterisk-users] rxfax, spandsp and lack of ecm
Artifex Maximus
- [asterisk-users] Re: PRI: sometimes Asterisk drop calls
Tomislav Parčina
- [asterisk-users] Calls on hold
Mir
- [asterisk-users] IVR not able to Play the Balance.. need some help
here
ram
- [asterisk-users] Adding own info in AMI
Mir
- [asterisk-users] (no subject)
Panagiotis Zikos
- [asterisk-users] Asterisk and "305 Use Proxy"
Wolfgang Hottgenroth
- [asterisk-users] voicemailmain errors on CLI
Benjamin Jacob
- [asterisk-users] Queue - static members
Tomislav Parčina
- [asterisk-users] Problem withfeature introducer as first digit on a
call
John covici
- [asterisk-users] HFC isdn card and bristuff 0.2.0 rc8n
Giordano Grandis
- [asterisk-users] Problems with call parking
Arik Raffael Funke
- [asterisk-users] sample configuration
Panagiotis Zikos
- [asterisk-users] Customize host in INVITE's Contact header?
Yoann Aubineau
- [asterisk-users] Streaming MoH Problem,
starts and then stops immediately
Zeeshan Zakaria
- [asterisk-users] ip address incoming call
antonio
- [asterisk-users] audio drop out half channel
Jerry Geis
- [asterisk-users] set global variable
Jan Fousek
- [asterisk-users] Building Zaptel 1.2.9 with Octasic
Mark Hulber
- [asterisk-users] Building Zaptel 1.2.9 with Octasic
Steven Totaro
- [asterisk-users] callback without agi
Patricio Valarezo
- [asterisk-users] University switches to Asterisk
Doug Lytle
- [asterisk-users] Third Lane PBX Manger Multi-Tenant
Bill Gibbs
- [asterisk-users] Dropped Calls on TDM400p
Jonathan Barratt
- [asterisk-users] Dropped Calls on TDM400p
Steven Totaro
- [asterisk-users] Asterisk crashing when monitoring SIP device with
Chanspy
mezzmor at aim.com
- [asterisk-users] success
Christopher Corn
- [asterisk-users] Unknown RTP codec 100 received
Thomas Artner
- [asterisk-users] Astmanproxy authentication problems
fho at inmedias.it
- [asterisk-users] Jitter Buffer on SIP
ggonzalez at telviso.com.ar
- [asterisk-users] Polycom 501 display bug
William M Conlon
- [asterisk-users] OT -- echo cancellation of an audio file
William M Conlon
- [asterisk-users] Snom 360 Function Keys
Conrad Wood
- [asterisk-users] Hangup on Panasonic KX-TEM824
ggonzalez at telviso.com.ar
- [asterisk-users] Polycom MyStat
Douglas Garstang
- [asterisk-users] Via Epia platforms and asterisk
Chris Bagnall
- [asterisk-users] Copyright issues with libcurl and OpenSSL
James Jones
- [asterisk-users] chan_zap.so stopped working after upgrading CentOS
Zeeshan Zakaria
- [asterisk-users] University switches to Asterisk
broadbandvoice at comcast.net
- [asterisk-users] Asterisk as a B2BUA and/or a FXS-SIP gateway?
AK 4asterisk
- [asterisk-users] QuadBRI and Zyxel Wifi phone stop working together
after 3 calls
Frederik Fix
- [asterisk-users] QuadBRI and Zyxel Wifi phone stop working
togetherafter 3 calls
Bill Gibbs
- [asterisk-users] polycom expansion module
Kevin Kiely
- [asterisk-users] How to install HUDLite Server
Zeeshan Zakaria
- [asterisk-users] IAX2 trunking
Siqhamo Sifo
- [asterisk-users] Flexible Wrap Up Time for Queue
Xue Liangliang
- [asterisk-users] changed behaviour of status indication on incoming
pri lines in asterisk 1.2?
Christian Mohrbacher
- [asterisk-users] urgently requires help regarding MWI
Tanzeel serfaraz
- [asterisk-users] Re: PRI: sometimes Asterisk drop calls
Tomislav Parčina
- [asterisk-users] One way audio problem on gateway to PSTN after
some time, no NAT involved
Kai Militzer
- [asterisk-users] RE: voicemailmain errors on CLI
Tomislav Parčina
- [asterisk-users] MWI problem on asterisk
Tanzeel serfaraz
- [asterisk-users] Re: Queue - static members
Tomislav Parčina
- [asterisk-users] Getting no Audio with G729
Benoît Mérouze
- [asterisk-users] ASTCC: change from no pin to pin request?
Ronald Wiplinger
- [asterisk-users] voicemail ,MWI problem
tanzeel sarfaraz
- [asterisk-users] DIAL and automatic/manual co line acces
Kai Ober
- [asterisk-users] Correct settings for UK (BT) FXO
Brian Candler
- [asterisk-users] SOLVED: ringback on box with E1 and premicell
yusuf
- [asterisk-users] Thomson 2030
Ricardo Carvalho
- [asterisk-users] Silence Call {very very urgent plz}
Abdul
- [asterisk-users] 9 becomes 99 ? And other strangeness
Brian Candler
- [asterisk-users] asterisk server to server using sip question
Jerry Geis
- [asterisk-users] voicemail access thru apache on another server
Benjamin Jacob
- [asterisk-users] VoiceXML browser for Asterisk available !
tech at i6net.com
- [asterisk-users] Dealing with FINAREA redirects
Arik Raffael Funke
- [asterisk-users] I need to record 30 conversations and have other
30 with music on hold, all at the same time, can a PIV handle it?
MF
- [asterisk-users] Asterisk and peers behind nat with port forward,
how to proxy?
Raul Dias
- [asterisk-users] Maximum retries exceeded on transmission
AJ Grinnell
- [asterisk-users] Attended Transfer Asterisk 1.2.11
Alberto Sagredo
- [asterisk-users] Detect PBX vs Network message
Savoy, Kevin - Williston, ND
- [asterisk-users] Getting 'i' functionality on internal extensions
Brian Candler
- [asterisk-users] sip show peers
Eric Rousse
- [asterisk-users] BLF across asterisk trunks
Norris, Sam
- [asterisk-users] Controlling the channel
Doug Lytle
- [asterisk-users] Forcing Marker bit, because SSRC has changed
Richard Klingler
- [asterisk-users] Disappearing Voicemail
Andrew Kirch
- [asterisk-users] WAIT FOR DIGIT not working
Joel Lansden
- [asterisk-users] sip show peers
Andrew Kirch
- [asterisk-users] asterisk server to server using sip question
Steven Totaro
- [asterisk-users] BLF across asterisk trunks
Douglas Garstang
- [asterisk-users] Incoming SIP provider goes unregistered and never
recovers
John Lingate
- [asterisk-users] incoming call h323 cdr
antonio
- [asterisk-users] Controlling the channel
Steven Totaro
- [asterisk-users] 9 becomes 99 ? And other strangeness
Steven Totaro
- [asterisk-users] Asterisk 1.4 Docs
Douglas Garstang
- [asterisk-users] 491 request pending
harrygaillac-sip at yahoo.fr
- [asterisk-users] Zork & Asterisk; zoip 0.2.0 released
Simon P. Ditner
- [asterisk-users] Page() paging application problem
Michael
- [asterisk-users] University switches to Asterisk
Steven Totaro
- [asterisk-users] Disappearing Voicemail
Andrew Kirch
- [asterisk-users] BLF across asterisk trunks
Norris, Sam
- [asterisk-users] [asterisk-dev] 491 request pending
harrygaillac-sip at yahoo.fr
- [asterisk-users] How to send DTMF down a channel
Frank Church
- [asterisk-users] (Off-topic) Voip number "tracert"
Daniel Cyt
- [asterisk-users] loosing Sipura 841 almost exactly on the hour
Jim Sturtevant
- [asterisk-users] OT, Definity G3 Problems with Asterisk (Any Avaya
Steven Totaro
- [asterisk-users] asterisk server to server using sip question
Jerry Geis
- [asterisk-users] how to transfer a caller out of a queue ?
Stefan-Michael. Guenther (in-put GbR)
- [asterisk-users] WAIT FOR DIGIT not working
Joel Lansden
- [asterisk-users] [asterisk-dev] 491 request pending
harrygaillac-sip at yahoo.fr
- [asterisk-users] mISDN versus ZapHFC with BRIstuff
Henrik Woffinden
- [asterisk-users] Sphinx2
Douglas Garstang
- [asterisk-users] Asterisk / Patton SmartNode SN2400 Strangeness FYI
George Pajari
- [asterisk-users] problems with Polycom 500 boot up
Forum
- [asterisk-users] G729 and Tribox
Bartosz Wegrzyn - asterisk
- [asterisk-users] Re: Detect PBX vs Network message
Savoy, Kevin - Williston, ND
- [asterisk-users] Sphinx2
Douglas Garstang
- [asterisk-users] Page() paging application problem
Michael
- [asterisk-users] Login user
Sergio R. D'Ippolito
- [asterisk-users] Asterisk with Addpac 2120
Pablo Almido
- [asterisk-users] about 'zap show channels'
PSPunch
- [asterisk-users] Why not g726-32?
RR
- [asterisk-users] Urgent !!!! Unable to make calls from cisco
callmanager to asterisk
Anantha Padmanabha.M.L
- [asterisk-users] ASTERISK HIGH AVAILABILITY
Giedrius Augys
- [asterisk-users] Cisco 79xx and vlan
Julian Lyndon-Smith
- [asterisk-users] How to download asterisk 1.3 development version
Boneym
- [Asterisk-Users] Can you explain why multiple registration is an
important (missing) feature ?
Olivier
- [asterisk-users] Cisco 7961 "dropouts"
Nat Wong
- [asterisk-users] Bri Card for Asterisk ?
Noc Phibee
- [asterisk-users] non-technical, dealing with users giving feedback
stoffell
- [asterisk-users] two safe_asterisk processes on the same PBX???
Giorgio Incantalupo
- [asterisk-users] Re: PRI: sometimes Asterisk drop calls
Tomislav Parčina
- [asterisk-users] Modem calls
Jose Limeres
- [asterisk-users] Re: Maximum retries exceeded on transmission
Tomislav Parčina
- [asterisk-users] Re: BLF across asterisk trunks
Tomislav Parčina
- [Asterisk-Users] Shared Line Appearance, Snom and trunk
Olivier
- [asterisk-users] RE: Asterisk 1.4 Docs
Tomislav Parčina
- [asterisk-users] AOC - advice of charge
Tomislav Parčina
- [asterisk-users] Re: Cisco 79xx and vlan
Tomislav Parčina
- [asterisk-users] Anyone using Voicemail with IMAP Support?
Arnd Vehling
- [asterisk-users] Re: Can you explain why multiple registration is
an important (missing) feature ?
Tomislav Parčina
- [asterisk-users] trying to understand siprealtime & nat/MWI issues
Arne Van Theemsche
- [asterisk-users] 491 request pending [2]
harrygaillac-sip at yahoo.fr
- [asterisk-users] Setting up imap based voicemail / invalid remote
specification
Arnd Vehling
- [asterisk-users] Modem calls
Jose Limeres
- [asterisk-users] CDR question with SIP/IAX trunks
Koopmann, Jan-Peter
- [asterisk-users] Compile error in Asterisk 1.2.12.1
Henrik Woffinden
- [asterisk-users] Cisco Distinctive ring using alert-info
Julian Lyndon-Smith
- [asterisk-users] Bristuffed asterisk 1.2.10 on Suse 10 - problem
with module versionmagic
Robert Rozman
- [asterisk-users] Section '12345678' lacks type
Lennart Utgård
- [asterisk-users] where download app_txfax?
Jerry Geis
- [asterisk-users] Issues with AGI+Dial command
Brian Rogan
- [asterisk-users] Sphinx2
Douglas Garstang
- [asterisk-users] Polycom 501 - message waiting LED manipulation
Mike
- [asterisk-users] open letter
harrygaillac-sip at yahoo.fr
- [asterisk-users] [asterisk-dev] open letter
harrygaillac-sip at yahoo.fr
- [asterisk-users] 4-wire analogue interfaces?
Tony Mountifield
- [asterisk-users] ZT_SPANCONFIG failed on span 1: No such device or
address (6)
Juan Miguel Yamakawa
- [asterisk-users] inbound call from GSM gateway:
handle_request_invite: Failed to authenticate user
Allan Kamau
- [asterisk-users] [OT] Nokia E60/61/70 and SIP
Martin Joseph
- [asterisk-users] Branch office interconnect - IAX :vs: SIP?
Gary G. Hendershot
- [asterisk-users] Cisco GW & CID Name
Peder at NetworkOblivion
- [asterisk-users] Internal message being heard on pstn line
Ryder Brook
- [asterisk-users] Asterisk with cisco 7935
Miles Scruggs
- [asterisk-users] Voicemail adjustments
Ricardo Carvalho
- [asterisk-users] app_txfax segv fault
Jerry Geis
- [asterisk-users] Has anyone tried to install both digital card and
analog card in one machine
Xue Liangliang
- [asterisk-users] Attended transfer and parking calls
Elpidio Ramos
- [asterisk-users] New astGUIclient VICIDIAL Release: 2.0.1
Matt Florell
- [asterisk-users] Reliability of the newer IAXy's
Lists
- [asterisk-users] Sphinx2
Douglas Garstang
- [asterisk-users] Asterisk variables
Mike
- [asterisk-users] Asterisk 1.2.12.1 and Zaptel 1.2.9.1 Released
The Asterisk Development Team
- [asterisk-users] blf aastra 9133i working but can't pickup calls
Gareth Owen
- [asterisk-users] Digium G.729 codec now available for Solaris/SPARC
Kevin P. Fleming
- [asterisk-users] FollowMe question
Hall, Eric M.
- [asterisk-users] DTMF Tone Not Passing Help
Nitesh Divecha
- [asterisk-users] pickupgroup 1
Ronald Wiplinger
- [asterisk-users] AEL2 patch for Asterisk 1.2.12.1
Barzilai
- [asterisk-users] Help spread the word about Asterisk!
Ronald Lewis
- [asterisk-users] voxee, callerid and trixbox
Christopher Corn
- [asterisk-users] saved.gsm -> Voicemail greeting ??
Nick Ellson
- [asterisk-users] amr codec
Net Nut
- [asterisk-users] call across 2 asterisks
unplug
- [asterisk-users] Scaling/Loadbalancing a Call Center and Redundancy
Steve Totaro
- [asterisk-users] saved.gsm -> Voicemail greeting ??
Bill Gibbs
- [asterisk-users] saved.gsm -> Voicemail greeting ??
Bill Gibbs
- [asterisk-users] saved.gsm -> Voicemail greeting ??
Bill Gibbs
- [asterisk-users] Asterisk as a gateway to SER
Siqhamo Sifo
- [asterisk-users] Integrating the Openser for VoiceMail and PBX with
Asterisk, For Account
raviprakash sunkara
- [asterisk-users] read variable from shell script
Christophorus Laube
- [asterisk-users] "Ping" a phone
Julian Lyndon-Smith
- [asterisk-users] SHSU asterisk installation?
Roy Sigurd Karlsbakk
- [asterisk-users] USA Regulatons
Marnus van Niekerk
- [asterisk-users] RE: [Asterisk-video] VXIasterisk is available !
Dean Collins
- [asterisk-users] Mediatrix 1204
Bill Michaelson
- [asterisk-users] Calling to PSTN newbie question
Pablo Almido
- [asterisk-users] Polycom programmable buttons
Ron McCarthy
- [asterisk-users] Wrong outgoing port
Master_PE
- [asterisk-users] system cmd
unplug
- [asterisk-users] How does Asterisk determine an incoming SIP
Channel name?
kjcsb
- [asterisk-users] Starting out
Timothy Parez
- [asterisk-users] IAX2 audio problem
Siqhamo Sifo
- [asterisk-users] Does a "HST Saphir III ML PCI" work with Asterisk?
Patrick Cervicek
- [asterisk-users] Asterisk Server Down
broadbandvoice at comcast.net
- [asterisk-users] Asterisk Server Down
Guido Hecken
- [asterisk-users] Asterisk Server Down
Guido Hecken
- [asterisk-users] A1200+fxo, anyone using this?
Nick Ellson
- [asterisk-users] RE: FollowMe question
Hall, Eric M.
- [asterisk-users] Register message received from realtime peer
crashes Asterisk
kjcsb
- [asterisk-users] Asterisk Server Down
broadbandvoice at comcast.net
- [asterisk-users] Termination Rates
broadbandvoice at comcast.net
- [asterisk-users] problem installing func_odbc on asterisk 1.2 ...
Vince Daneff
- [asterisk-users] Polycom Expansion Module
Douglas Garstang
- [asterisk-users] Noob question: Packet size
Avi Miller
- [asterisk-users] RE: Asterisk 1.4 Docs
Tomislav Parčina
- [asterisk-users] Re: Can you explain why multiple registration is
an important (missing) feature ?
Tomislav Parčina
- [asterisk-users] Xorcom Astribank
Klaus Darilion
- [asterisk-users] is chanisavail command reliable?
Giorgio Incantalupo
- [asterisk-users] Log out an Agent on RNA
mbodbg at gmx.net
- [asterisk-users] disconnect code in featuremap doesn't work on
unanswered calls
Stanley Cline
- [asterisk-users] Queue - Agent language
Tomislav Parčina
- [asterisk-users] Re: Log out an Agent on RNA
Tomislav Parčina
- [asterisk-users] pickup call little complicated
Miloš Kocbek
- [asterisk-users] Variable that gives the SIP channel
Andre Courchesne - Consultant
- [asterisk-users] Problem with Asterisk Realtime (MySQL)
Edwin Pauli
- [asterisk-users] Playtones
Tomislav Parčina
- [asterisk-users] User authentication
Siqhamo Sifo
- [asterisk-users] Asterisk Design Question
Remi Quezada
- [asterisk-users] Chanspy crashing the server, again
mezzmor at aim.com
- [asterisk-users] Re: Mediatrix 1204 trix
Bill Michaelson
- [asterisk-users] FOP Installation help
Zeeshan Zakaria
- [asterisk-users] RE : Re: [asterisk-dev] open letter
harrygaillac-sip at yahoo.fr
- [asterisk-users] unable to change the emailbody for email
notification
richard Coco
- [asterisk-users] Chanspy crashing the server, again
mezzmor at aim.com
- [asterisk-users] Polycom SoundPoint 2.0.1 SIP firmware?
Ken D'Ambrosio
- [asterisk-users] Chanspy crashing server, again
mezzmor at aim.com
- [asterisk-users] Cisco 7940 Problem (Mess)
Gary Guthary
- [asterisk-users] Dial and Timeout
Tobias Wolf
- [asterisk-users] LDAP athentication
Andre O.
- [asterisk-users] ASTERFAX
Scott Pinhorne
- [asterisk-users] Changes in extensions.conf handling between 1.2 &
1.4
Dave Cotton
- [asterisk-users] X100P and zaptel 1.2.8
Tim
- [asterisk-users] How to make Polycom 501 go off hook when pressing
any digits
Mike
- [asterisk-users] Fedora
bilal ghayyad
- [asterisk-users] CSR introduces UniVox reference platform
Steve Kennedy
- [asterisk-users] how to transfer a caller out of a queue ?
Stefan-Michael. Guenther (in-put GbR)
- [asterisk-users] INSTALL_PREFIX=
Forrest Beck
- [asterisk-users] Asterisk / Audiocodes annoying issue - Seeking
Suggestions
Cory Andrews
- [asterisk-users] ANI and Meetme...
Natambu Obleton
- [asterisk-users] sip.conf for talking to other Asterisk machines
Bill Gibbs
- [asterisk-users] Fedora
Guido Hecken
- [asterisk-users] Periodic announcements & MySQL Realtime
Sébastien Mortier
- [Asterisk-Users] How to learn or teach VoIP QoE
Olivier
- [asterisk-users] Digium GUI?
shadowym
- [asterisk-users] Digium GUI?
Don Fanning
- [asterisk-users] Enabling Second Processor Trashes Audio Quality
George Pajari
- [asterisk-users] Asterisk Appliance,
will Asterisk Business Edition be mandatory?
shadowym
- [asterisk-users] Starting Asterisk PBX: FATAL: Module ixj not found.
broadbandvoice at comcast.net
- [asterisk-users] create_addr: No such host:
broadbandvoice at comcast.net
- [asterisk-users] Starting Asterisk PBX: FATAL: Module ixj not
found.
broadbandvoice at comcast.net
- [asterisk-users] Termination Rates
Insider KT
- [asterisk-users] Termination Rates
broadbandvoice at comcast.net
- [asterisk-users] Polycom SoundPoint 2.0.1 SIP firmware?
Douglas Garstang
- [asterisk-users] spandsp fax using Asterisk 1.2.X
isamar at isamarmaia.org
- [asterisk-users] Polycom Expansion Module
Douglas Garstang
- [asterisk-users] Polycom programmable buttons
Douglas Garstang
- [asterisk-users] Digium GUI?
Douglas Garstang
- [asterisk-users] sip.conf for talking to other Asterisk machines
Douglas Garstang
- [asterisk-users] Accounting and re-invite
Ronald Wiplinger
- [asterisk-users] 488 Not acceptable here sent by Asterisk - SIP
debug follows
Dinesh Nair
- [asterisk-users] Re: unable to change the emailbody for email
notification
Tomislav Parčina
- [asterisk-users] Query on MWI
Tanzeel serfaraz
- [asterisk-users] prompt playing problem
unplug
- [asterisk-users] Re: Playtones
Tomislav Parčina
- [asterisk-users] Leave Queue when all agents busy
Mindaugas Kezys
- [Asterisk-Users] CallerID retain on internal transfer
Olivier
- [asterisk-users] How to Dial a number with Sangoma PRI card?
Mario
- [asterisk-users] Wrong call handling
Erik Wartusch
- [asterisk-users] When does Scalability requests Asterisk to Use SER
?
Marco Mouta
- [asterisk-users] Alcatel OXO Sip
Christian Gatti
- [asterisk-users] Alcatel OXO Sip
Steve Langstaff
- *****SPAM***** [asterisk-users] iax2 trunk call limits
Ma Zhiyong
- [asterisk-users] Anyone Using a Patton (Inalp) SmartNode 2400 for
T.38?
George Pajari
- [asterisk-users] Query ,NEED help regarding MWI
Tanzeel serfaraz
- [asterisk-users] Clustering architecture and echo cancellation issue
Panagiotis Zikos
- [asterisk-users] Aastra 9133i and Atcom AT-320 - Comments please
James Dyer
- [asterisk-users] Format_MP3, Streaming, File Formats, MOH
Robert Chadwell
- [asterisk-users] polycom 501 digitmap
Jordan Novak
- [asterisk-users] Problem with # locking up call
marvin horst
- [asterisk-users] fast SIP failover (outgoing sIP requests) with 1.2
Klaus Darilion
- [asterisk-users] fast SIP failover (outgoing sIP requests) wi
th 1.2
Guido Hecken
- [asterisk-users] transcoding error?
Damon Estep
- [asterisk-users] Digium GUI?
Douglas Garstang
- [asterisk-users] fast SIP failover (outgoing sIP requests) with
1.2
Damon Estep
- [asterisk-users] Polycom default handset volume
Damon Estep
- [asterisk-users] When does Scalability requests Asterisk
AK 4asterisk
- [asterisk-users] Semi-OT: SIP or IAX provider in the Boston area?
Erik Anderson
- [asterisk-users] When does Scalability requests Asterisk
AK 4asterisk
- [asterisk-users] Call forward with CFU?
Roy Sigurd Karlsbakk
- [asterisk-users] Polycom default handset volume
Douglas Garstang
- [asterisk-users] mpg123
Giordano Grandis
- [asterisk-users] codecs/voicemail/DTMF
Mr. Jones
- [asterisk-users] 404 not Found
harrygaillac-sip at yahoo.fr
- [asterisk-users] SIP "Lines" Example Citel
Steve Totaro
- [asterisk-users] Repost: Register message received from realtime
peer crashes Asterisk
Cameron and Karlene
- [asterisk-users] gTalk no audio issue
Alex Guan
- [asterisk-users] grandstream gxp 2000 does not display names when
calling out
Christopher Corn
- [asterisk-users] Pri Event 6 and 8
Dave Wise
- [asterisk-users] SPA 3102 does not even attempt to register
Allan Kamau
- [asterisk-users] clustering asterisk is possible ?
Información Capa Tres
- [asterisk-users] Asterisk AGI question
David R.
- [asterisk-users] DTMF Detection Problems with certain phones
incoming zap channels
asterisk asterisk
- [asterisk-users] Grandstream SX2000 attended tranfer
magnus
- [asterisk-users] g729 and polycoms problem
Delca
- [asterisk-users] g729 and polycoms problem
Alyed Tzompa
- [asterisk-users] IAX or SIP termination provider that reaches
6421xxxxxxx?
Mojo with Horan & Company, LLC
- [asterisk-users] Cisco 7960 part numbers ...
Cesc
- [asterisk-users] IAX or SIP termination provider that reaches
6421xxxxxxx?
Mojo with Horan & Company, LLC
- [asterisk-users] Polycom 500 power supply
Forum
- [asterisk-users] MOH distorted on Pound Key Linux on asterisk 1.2.8
Jeronimo Romero
- [asterisk-users] SkypeOut with Asterisk?
Devraj Mukherjee
- [asterisk-users] Re: Playtones
Tomislav Parčina
- [asterisk-users] Uninstalling Trixbox
Rizwan Hisham
- [asterisk-users] Re: mpg123
Tomislav Parčina
- [asterisk-users] stress a server with a tool
nik600
- [asterisk-users] BRI: Asterisk disconnecting on 'call diverted'
message?
Benoit Panizzon
- [asterisk-users] Channel kept busy when creating ssh tunnel via AGI
Giorgio Incantalupo
- [asterisk-users] agi
Gopal krishnan
- [asterisk-users] Mediant 1000
Rajkumar S
- [asterisk-users] Registration doubt
god
- [asterisk-users] Register doubt
god
- [asterisk-users] University dumps CISCO VoIP for Asterisk
Ferguson, Michael
- [asterisk-users] enumlookup - deprecated working - but appreciated
one duznt :-(
Benjamin Jacob
- [asterisk-users] Digium GUI?
Douglas Garstang
- [asterisk-users] Forwarding the Ring Group and Calls coming in to
Queues
Zeeshan Zakaria
- [asterisk-users] Incoming calls, identify
joea, j4computers
- [asterisk-users] University dumps CISCO VoIP for Asterisk
joea, j4computers
- [asterisk-users] Realtime madness
Scott Pinhorne
- [asterisk-users] MOH distorted on Pound Key Linux on asterisk1.2.8
Jeronimo Romero
- [asterisk-users] Sip configuration using mysql
Arkaitz
- [asterisk-users] tx_fax over sip to TDM card
Jerry Geis
- [asterisk-users] MOH distorted on Pound Key Linux on asterisk1.2.8
Jeronimo Romero
- [asterisk-users] Unexpected delay
flavio
- Asterisk capabilities, was [asterisk-users] University dumps CISCO
VoIP for Asterisk
joea, j4computers
- [asterisk-users] Zap channel digit.
Luca Salemmi
- [asterisk-users] Available channels
Steve Kennedy
- [asterisk-users] How to register from asterisk server to an xlite.
god
- [asterisk-users] (no subject)
asterisk at teladdict.com
- [asterisk-users] HINT problems with SVN-trunk-r43322
Hall, Eric M.
- [asterisk-users] No channels available after reloading config
Richard Klingler
- [asterisk-users] Cannot hear the other side of the phone call
Dennis P. Clark
- [asterisk-users] A-Z termination
asterisk at teladdict.com
- [asterisk-users] Getting Music On Hold working in * 1.2.12.1 with
Fedora?
voiplist
- [asterisk-users] Re: Uninstalling Trixbox
Nic Hughes
- [asterisk-users] Asteisk plays music on hold starting from random
point
Giorgio Incantalupo
- [asterisk-users] HINT problems with SVN-trunk-r43322
Watkins, Bradley
- [asterisk-users] Asterisk Voicemail with Sonus?
Chris Carey
- [asterisk-users] SPA-3102 PSTN->VoIP Gateway (quest for one stage
dialing)
Allan Kamau
- [asterisk-users] RTT in rtcp debug
André Abrantes
- [asterisk-users] Round Robin + Ringall
equis software
- [asterisk-users] ztdummy installed but choppy audio warning
Juanjo Portela
- [asterisk-users] PRI Backup
Forrest Beck
- [asterisk-users] Cisco 7970 behind NAT
Jeremiah Millay
- [asterisk-users] A Caller ID question (UK)
Gordon Henderson
- [asterisk-users] macro-dialout-trunk without agi or manager
Nathan Bell
- [asterisk-users] Re: Can you explain why multiple
registration isan important (missing) feature
Tielin Xu
- [asterisk-users] g729 and polycoms problem
Alyed Tzompa
- [asterisk-users] Asterisk Bussiness Edition and Realtime.
Aquiles Cohen
- [asterisk-users] Cisco 7970 behind NAT
Alyed Tzompa
- [asterisk-users] voice detection during playback
David Koski
- [asterisk-users] Incoming calls, identify
joea, j4computers
- [Asterisk-Users] No Sound from VoicemailMain - Device Linksys PAP2T-NA
Tracy Lofton
- [asterisk-users] X100P compile problems
Tim
- [asterisk-users] problems with Polycom 500 boot up
Dean Collins
- [asterisk-users] Configuring Codecs
wyatt.wmvg at comcast.net
- [asterisk-users] University dumps CISCO VoIP for Asterisk
Douglas Garstang
- [asterisk-users] Configuring Codecs
Alyed Tzompa
- [asterisk-users] Polycom 2.0.1 Software
Douglas Garstang
- [asterisk-users] Call/Voicemail Screening
Lacy Moore - Aspendora
- [asterisk-users] Polycom 2.0.1 Software
Douglas Garstang
- [asterisk-users] Re: Can you explain why multiple registration isan
important (missing) feature ?
Tomislav Parčina
- [asterisk-users] Help in Reloading of Asterisk...
raviprakash sunkara
- [asterisk-users] RTCP and RTP packetization in 1.4
yusuf
- [asterisk-users] How much SIP calls can I squeeze from this box
Erick Perez
- [asterisk-users] Unexpected delay: problem with outgoing calls
flavio
- [asterisk-users] Iax Netstat Output
Arun Kumar
- [asterisk-users] Habitual set of number
Vitaly Oborsky
- [asterisk-users] asterisk skills in the philippines
tubongpeyups
- [asterisk-users] asterisk / chan_capi problems
Klaus Darilion
- [asterisk-users] Invite issues
Allan Kamau
- [asterisk-users] HINT problems with SVN-trunk-r43322
Watkins, Bradley
- [asterisk-users] Two phones, same number
Benny Amorsen
- [asterisk-users] asterisk, iaxmodem, hylafax quality problem
Artifex Maximus
- [asterisk-users] ChanIsAvail
Steve Kennedy
- [asterisk-users] Using Asterisk with IVR connected with legacy pbx
via rs-232
Paulo Garcia
- [asterisk-users] VoicemailMain()
Michel Zenone
- [Asterisk-users] Calls between IAX2 Clients don't work correctly
Llorenç Suau
- [asterisk-users] Setting QOS settings in asterisk and/or CentOS?
BerkHolz, Steven
- [asterisk-users] Looped message playback
Earle Clubb
- [asterisk-users] Asteisk plays music on hold starting from
randompoint
Robert Chadwell
- [asterisk-users] Format_MP3, Streaming, File Formats, MOH
Robert Chadwell
- [asterisk-users] Format_MP3, Streaming, File Formats, MOH
Robert Chadwell
- [asterisk-users] Cisco 7970 behind NAT
Jeremiah Millay
- [asterisk-users] Looped message playback
Bill Gibbs
- [asterisk-users] Looped message playback
Bill Gibbs
- [asterisk-users] PRI Problems
Klaus Darilion
- [asterisk-users] Format_MP3, Streaming, File Formats, MOH
Frédéric Marti
- [asterisk-users] Setting QOS settings in asterisk and/or CentOS?
Redouane Doumer
- [asterisk-users] Re: Uninstalling Trixbox
Nic Hughes
- [asterisk-users] CURL
Elpidio Ramos
- [asterisk-users] CURL
Jerry Geis
- [asterisk-users] TDM2400P
Robson Ribeiro
- [asterisk-users] notransfer local channel on redirect
Benko
- [asterisk-users] Call is dead after featuredigittimeout
Florian Hars
- [asterisk-users] Polycom 650 Question
Dean Collins
- [asterisk-users] Linksys SPA400
Carlos Chavez
- [asterisk-users] Asterisk and Panasonic D500
Carlos Chavez
- [asterisk-users] RE: Setting QOS settings in asterisk and/or CentOS?
Greg Kennedy
- [asterisk-users] Asteisk plays music on hold
startingfrom randompoint
Robert Chadwell
- [asterisk-users] Format_MP3, Streaming, File Formats, MOH
Robert Chadwell
- [asterisk-users] TDM2400 wired description and skiping frames
Robson Ribeiro
- [asterisk-users] CURL
Colin Anderson
- [asterisk-users] asterisk and PowerEdge 1950
Stas Khromoy
- [asterisk-users] mISDN problem: no version for "capi_cmd2str" found
Arik Raffael Funke
- [asterisk-users] Asterisk 1.4 Beta Uploaded!!!
Matt Riddell (IT)
- [asterisk-users] Mini call center only 15 seats fxs to sip
suggestion
Erick Perez
- [asterisk-users] Asterisk, Asterisk-Addons,
Zaptel and Libpri 1.4 betas released!
Asterisk Development Team
- [asterisk-users] SPA941 -> Asterisk -> Voip provider -> PSTN ->
ShoreTel garble
Cliff Brake
- [asterisk-users] Hardware question
Jeremy Jongsma
- [asterisk-users] Extensions problems
equis software
- [asterisk-users] IAX or SIP termination provider that
reaches6421xxxxxxx?
Dean Collins
- [asterisk-users] TDM2400P
Jordan Novak
- [asterisk-users] DSL router with integrated SIP proxy?
Brian Candler
- [asterisk-users] g729 and polycoms problem
Alyed Tzompa
- [asterisk-users] Dual asterisk, CallerID(name|number) problem
Delca
- [asterisk-users] CURL
Colin Anderson
- [asterisk-users] Integrating Asterisk with LDAP Realtime
Nick Couchman
- [asterisk-users] TDM2400 problem isolated with POLYCOM IP301
phones!!!
Robson Ribeiro
- [asterisk-users] Any one knows if I can connect SS7 to Asterisk,
(with a TE405P or other) ???
MF
- [asterisk-users] SOS building fastagi C
Yelson Vivas
- [asterisk-users] iaxyprov downloading problems
Sean Kennedy
- [asterisk-users] zttest output
Jason
- [asterisk-users] multiple zaptel cards
Jordan Novak
- [asterisk-users] iaxy configuration problems
Sean Kennedy
- [asterisk-users] Application of Asterisk Packetization Patch
pannel at optusnet.com.au
- [asterisk-users] Dynamic DNS asterisk server?
Martin Joseph
- [asterisk-users] Application of Asterisk Packetization Patch
Dan Austin
- [asterisk-users] Re: Can you explain why multiple registration isan
important (missing) feature ?
Tomislav Parčina
- [asterisk-users] Picking up a call from queue?
Rajkumar S
- [asterisk-users] Fax Detection on outbound call
Mark Edwards
- [asterisk-users] Format_MP3, Streaming, File Formats, MOH
Frédéric Marti
- [asterisk-users] Dual core
Tomislav Parčina
- [asterisk-users] E1 - PCI-Express
Tomislav Parčina
- [asterisk-users] new in 1.4?
Roy Sigurd Karlsbakk
- [asterisk-users] alternatives to mpg123: format_mp3,
rawplayer or madplay?
Giorgio Incantalupo
- [asterisk-users] Iax2 show netstat
Arun Kumar
- [asterisk-users] INVITE re-try interval
David Brazier
- [asterisk-users] ATA with wireless client
Brian Candler
- [asterisk-users] freepbx dial plan, add and remove at the same time
Mike Williams
- [asterisk-users] Iax Netstat Output
Arun Kumar
- [Asterisk-Users] Where to find error codes
Olivier
- [asterisk-users] Does Asterisk 1.4 going to support realtime
ex-girlfriend logic?
Ricardo Carvalho
- [asterisk-users] How can the User Know he has voicemail in the
Databases.
raviprakash sunkara
- [asterisk-users] Polycom phone references needed
AstATN
- [asterisk-users] Question about SVN-trunk-r43322 and Asterisk
Recording Interface
Hall, Eric M.
- [asterisk-users] Asterisk ramdonly crash using Realtime Static
equis software
- [asterisk-users] Re: [asterisk-dev] To bweschke regarding app
FollowMe
BJ Weschke
- [asterisk-users] Help with Tieing Outbound calls to Zap Channels
Kevin Steil
- [asterisk-users] Callback
Khaled Chehab
- [asterisk-users] hint status from dialplan?
BerkHolz, Steven
- [asterisk-users] Re: [asterisk-dev] To bweschke regarding app
FollowMe
BJ Weschke
- [asterisk-users] experience with phones locking up uniden and cisco
Jerry Geis
- [Asterisk-Users] UK Male English Voices
Will Tatam
- [Asterisk-Users] Re: [asterisk-biz] UK Male English Voices
Will Tatam
- [asterisk-users] Forcing Marker bit, because SSRC has changed
Richard Klingler
- [asterisk-users] dialout-trunk vs. dial group
Nathan Bell
- [asterisk-users] Asterisk & MSN ?
Yoann Aubineau
- [asterisk-users] 64 analog phones
mike
- [asterisk-users] Display message on voip phone...hint?
Ale
- [asterisk-users] chan_isdn / chan_sip problems
Arik Raffael Funke
- [asterisk-users] Re: [asterisk‑users] Integrating Asterisk with LDAP Realtime
Nick Couchman
- [asterisk-users] Display message on voip phone...hint?
Colin Anderson
- [asterisk-users] SNOM 320 - 404 "Not Found"
oi geli
- [asterisk-users] Asterisk 1.4-beta2 Spanish Sounds missing
vm-youhaveno?
Earl Terwilliger
- [asterisk-users] ZAP: psuedo camped on channel 1?
Barry D. Hassler
- [asterisk-users] channel.c: Nobody there, continuing...
Chris Miller
- [asterisk-users] SNOM 320 - 404 "Not Found"
Christian Stredicke
- [asterisk-users] Digium G.729 codec binaries updated for Asterisk
1.4 beta
Jason Parker
- [asterisk-users] Question about SVN-trunk-r43322 and Asterisk
Recording Interface
Hall, Eric M.
- [asterisk-users] Very high ping times from 7960 phones
Steve Glaus
- [asterisk-users] Polycom (and others) digitmap info
Jerry Jones
- [asterisk-users] Leased line interconnect
Eric Bishop
- [asterisk-users] OT: Anybody remember this from last Dec?
Colin Anderson
- [asterisk-users] make error
joea, j4computers
- [asterisk-users] Comments on new system plan.
Dave Fullerton
- [asterisk-users] iaxy will register, but doesn't detect POTS line
Sean Kennedy
- [asterisk-users] OT But So Ungodly Important
Rushowr
- [asterisk-users] Polycom phone help needed
AstATN
- [asterisk-users] Trixbox Documentation
Rizwan Hisham
- [asterisk-users] Trixbox Documentation
joea, j4computers
- [asterisk-users] 1.4 Beta 2 Config Problem
Keith O'Brien
- [asterisk-users] AGI Errors
leitstelle at bnet.li
- [asterisk-users] make error
joea, j4computers
- [asterisk-users] libpri current extracts as beta1
joea, j4computers
- [asterisk-users] Connecting Motorola VT2442 Device
David Evennou
- [asterisk-users] One server SUBSCRIBE for information on multiple
voicemail boxes
Chris Carey
- [asterisk-users] Debugging and Outbound SIP Trunk
wyatt.wmvg at comcast.net
- [asterisk-users] Conference Call Delay & Quality
wyatt.wmvg at comcast.net
- [asterisk-users] Problem with zaptel 1.4b2 and X101P Wildcard
Morten Isaksen
- [asterisk-users] lumenvox speech recognition
Dean Collins
- [asterisk-users] Trixbox Documentation
joea, j4computers
- [asterisk-users] libpri current extracts as beta1
joea, j4computers
- [asterisk-users] Dual core
Bill Gibbs
- [asterisk-users] Integrating Asterisk with LDAP Realtime
Nick Couchman
- [asterisk-users] RE: 1.4 Beta 2 Config Problem
Keith O'Brien
- [asterisk-users] fax over ip
Christopher Corn
- [asterisk-users] e911
Christopher Corn
- [asterisk-users] Segmentation fault on Asterisk startup:
res_config_mysql.so problem?
kjcsb
- [asterisk-users] Cisco 7960 Double Natted
Barry Fawthrop
- [asterisk-users] running ooh323 on asterisk-1.14beta2
Илья Шипицин
- [hylafax-users] [asterisk-users] asterisk, iaxmodem,
hylafax quality problem
Artifex Maximus
- [asterisk-users] dialplan for confrencing
imthiyaz at peopletech.co.in
- [asterisk-users] Asterisk+Astbill
Siqhamo Sifo
- [asterisk-users] RE: 1.4 Beta 2 Config Problem
Keith O'Brien
- [asterisk-users] RE: 1.4 Beta 2 Config Problem
Keith O'Brien
- [asterisk-users] iaxy: one way audio
Sean Kennedy
- [asterisk-users] Missing sound in spanish from 1.4 beta2
Carlos Chavez
- [asterisk-users] 2 CPU's, Only 1 taking IRQ's
Bart Fisher
- [asterisk-users] how to configure a sip service
Raphael Jacquot
- [asterisk-users] Re: 2 CPU's, Only 1 taking IRQ's
Álvaro Palma
- [asterisk-users] spandsp (foip)
Christopher Corn
- [asterisk-users] Rpath PoundKey 1.2
Bobby Lee
- [asterisk-users] Need a recommended T38 FOIP solution
Christopher Corn
- [asterisk-users] High utilization with SIP registration
sdallan at ozemail.com.au
- [asterisk-users] RE: Dual core
Tomislav Parčina
- [asterisk-users] Re: Dual core
Tomislav Parčina
- [asterisk-users] Re: Dual core
Tomislav Parčina
- [asterisk-users] Re: Very high ping times from 7960 phones
Tomislav Parčina
- [asterisk-users] Snom MWI not turning off when message picked up.
Mike Dent
- [asterisk-users] ougoing calls problem
flavio
- [asterisk-users] AgentCallbacklogin in Asterisk1.4 beta2
Li yuqian
- [asterisk-users] voicemail greeting
unplug
- [asterisk-users] A Strange doubt and problem
Crazy Boy
- [asterisk-users] Cisco 7970 - DTMF
Tomislav Parčina
- [asterisk-users] Line Pickup Problem
Mohsen Basirat
- [asterisk-users] Call back
Khaled Chehab
- [asterisk-users] voicemail greeting
wyatt.wmvg at comcast.net
- [asterisk-users] Asterisk Trunk with Alcatel 4200 PABX
Frederico Madeira
- [asterisk-users] progress problems from SIP to PRI
Klaus Darilion
- [asterisk-users] ztcfg / X100P question
Michel Vaillancourt
- [asterisk-users] Re: Cisco 7970 - DTMF
Tomislav Parčina
- [asterisk-users] OT: Opinions on Aastra 480i CT?
Colin Anderson
- [asterisk-users] Queue failover and wrap time
Michelle Dupuis
- [asterisk-users] asterisk to cell phone network
yrving rivas
- [asterisk-users] REQUERIMIENTOS TE110P Y PANASONIC TDA620
DiegoF
- [asterisk-users] AGI Errors
leitstelle at bnet.li
- [asterisk-users] OT: Opinions on Aastra 480i CT?
Colin Anderson
- [asterisk-users] OT: Opinions on Aastra 480i CT?
Colin Anderson
- [asterisk-users] Cisco 7960 Double Natted
Hughes, Sam
- [asterisk-users] Running Multiple Instances of Asterisk
Douglas Garstang
- [asterisk-users] PBX TDA620 AND TE110P
DiegoF
- [asterisk-users] DUNDi Servers
Douglas Garstang
- [asterisk-users] Got SIP response 415 "Unacceptable Content-Type"
back from 192.168.1.209
Mr. Jones
- [asterisk-users] Asterisk 1.4 autoconf and /etc/asterisk directory
Douglas Garstang
- [asterisk-users] can someone recommened a reliable,
cheap t38 origination/termination provider
Christopher Corn
- [asterisk-users] Running Multiple Instances of Asterisk
Douglas Garstang
- [asterisk-users] channel.c: Nobody there, continuing...
Chris Miller
- [asterisk-users] Running Multiple Instances of Asterisk
Douglas Garstang
- [asterisk-users] trixbox t38 pass through
Christopher Corn
- [asterisk-users] TDM2400P vs Sangoma A200
Dave Fullerton
- [asterisk-users] Running Multiple Instances of Asterisk
Douglas Garstang
- [asterisk-users] How to stream audio to external app for speech
recognition and recognize dtmf in parallel ?
Robert Rozman
- [asterisk-users] include "context"
Eric Rousse
- [Asterisk-Users] Network impairment tools
Olivier
- [asterisk-users] Extensions busy/congested and "circuit-busy"
Guy M Guyadeen
- [asterisk-users] Shared Line Appearances in 1.4
Douglas Garstang
- [asterisk-users] MOH in 1.4 - Still Broken?
Douglas Garstang
- [asterisk-users] MOH in 1.4 - Still Broken?
Douglas Garstang
- [asterisk-users] fw: Uniden - TVUNIDEN_UIP300
Alyed Tzompa
- [asterisk-users] "does /var/run/asterisk.ctl exist?" -- but
Asterisk *is* running.
Ken D'Ambrosio
- [asterisk-users] MOH in 1.4 - Still Broken?
Douglas Garstang
- [asterisk-users] rtc: lost some interrupts at 1024 when loading
ztdummy
Geoff Karl
- [asterisk-users] Ericsson MD110
Patricio A. Bruna
- [asterisk-users] Unrecognized frames
Chris Miller
- [asterisk-users] RE: Dual core
Tomislav Parčina
- [asterisk-users] Re: Dual core
Tomislav Parčina
- [asterisk-users] Changing the recording dir of MeetMe recordings.
Jan du Toit
- [Asterisk-Users] The best way to track no-audio calls
Olivier
- [asterisk-users] Asterisk and UMTS phones
Andrea Spadaccini
- [asterisk-users] Re: asterisk to cell phone network
Tomislav Parčina
- [asterisk-users] core dump with 1.2.7.1 and chan-capi-cm 0.6.5
Klaus Darilion
- [asterisk-users] Cannot make outbound calls to some numbers with
PRI line from legacy PBX thru Asterisk with Sangoma
Giorgio Incantalupo
- [asterisk-users] asterisk - alcatel
et pourquoi pas ? epp
- [asterisk-users] error 407 authenticate on INVITE
unplug
- [asterisk-users] asterisk 1.4 branch and chan-capi-0.7.0
Guido Hecken
- [asterisk-users] asterisk 1.4 branch and chan-capi-0.7.0
Guido Hecken
- [asterisk-users] Call back
Khaled Chehab
- [asterisk-users] How can I stop lost DNS from killing Asterisk?
Eric Bishop
- [asterisk-users] Asterisk 1.4 mohsuggest
Douglas Garstang
- [asterisk-users] Play wav file during conversation
Eric
- [asterisk-users] Is there T.38 support on asterisk 1.4 beta2 ???
Ricardo Martins
- [asterisk-users] X100P Clone card in JAPAN
Miroslav Spasovski
- [asterisk-users] Asterisk behind Sonicwall firewall
Barry Fawthrop
- [asterisk-users] PRI Outbound CallerID Question
Shawn Kelley
- [asterisk-users] Included context
Artifex Maximus
- [asterisk-users] SIP Gateway
Forrest Beck
- [asterisk-users] PRI Outbound CallerID Question
Colin Anderson
- [asterisk-users] H323 IP phones
Alyed Tzompa
- [asterisk-users] Rewriting CID number w/o changing CDR src field
Mike Diehl
- [asterisk-users] voicemailmain menu
Jack Wei
- [asterisk-users] TE406P not working on Intel D101Ggc motherboard.
Álvaro Palma
- [asterisk-users] Asterisk 1.4 mohsuggest
Douglas Garstang
- [asterisk-users] IAX2 & SIP Monitoring Solution for Asterisk
David Thomas
- [asterisk-users] speaker phone echo
Christopher Corn
- [asterisk-users] mISDN, 2 Billion HFC ISDN cards,
cannot dial or receive
Nigel Godfrey
- [asterisk-users] Re: IAX2 & SIP Monitoring Solution for Asterisk
jaw+asterisk at tcp4me.com
- [asterisk-users] speaker phone echo
Colin Anderson
- [asterisk-users] señalizacion te110p, signaling te110p
DiegoF
- [asterisk-users] Problem with "Background" DTMF detection with A200D
Alvin Austin
- [asterisk-users] Context default & incoming ENUM
Ronald Wiplinger
- [asterisk-users] Grandstream GXV 3000
Chris HARIGA
- [asterisk-users] Priority "n"
Ronald Wiplinger
- [asterisk-users] How to change pager notification message
Mike Diehl
- [asterisk-users] max number of devices in hint
Lacy Moore - Aspendora
- [asterisk-users] queue information
unplug
- [asterisk-users] Anybody have the opvx1200.c driver?
Nick Ellson
- [asterisk-users] WARNING: chan_sip.c add_realm_authentication: ???
Ronald Wiplinger
- [asterisk-users] my (SIP) INVITE is ignored
lokotes
- [asterisk-users] outgoing call problem
Alexandru Voinescu
- [asterisk-users] Voip Buster - CID
Tomislav Parčina
- [asterisk-users] Configuring Asterisk 1.4-beta2 to work with jingle
Raffaele Porzio
- [asterisk-users] How can I unistall Asterisk?
Raffaele Porzio
- [asterisk-users] Re: max number of devices in hint
Tomislav Parčina
- [asterisk-users] Re: Advice of charge
Tomislav Parčina
- [asterisk-users] High CPU usage when Internet goes down
sdallan at ozemail.com.au
- [asterisk-users] ISDN30 and digital phones
Mike Williams
- [asterisk-users] T1 timing errors (Frame Slips) on Nortel 61C to
TE110P
Ronnie Jones
- [asterisk-users] Right way to prevent analog channel from answering
the phone?
Nick Ellson
- [asterisk-users] 64 analog phones
Bill Gibbs
- [asterisk-users] Good Book on Asterisk
Norbert Zawodsky
- [asterisk-users] Outgoing DialPlan
Scott Pinhorne
- [asterisk-users] IAX phones?
Ken D'Ambrosio
- [asterisk-users] TDM04B Installation Problem
Ian Chilton
- [asterisk-users] SER with multiple asterisk deployment
Adi Simon
- [asterisk-users] ASTTAPI
Mike Hammett
- [asterisk-users] 64 analog phones
Colin Anderson
- [asterisk-users] Zapata.conf
Danko Miocevic
- [asterisk-users] Queue Status via Dialplan
Rick Smith
- FW: [asterisk-users] Re: asterisk to cell phone network
Sam Tam
- [asterisk-users] Any suggestions about VoIP provider?
Antoine Megalla
- [asterisk-users] 64 analog phones
Colin Anderson
- [asterisk-users] Linksys/Sipura 3K, Calls Timing Out
Iain Young
- [asterisk-users] txfax question
Jerry Geis
- [asterisk-users] Cisco ATA escaping # into %23?
Scott Call
- [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P
Ronnie Jones
- [asterisk-users] TDM400P Problem or Not?
Eddie Johnson Jr
- [asterisk-users] AGI Errors
leitstelle at bnet.li
- [asterisk-users] RPID
DANIEL, AARON MATTHEW
- [asterisk-users] Spurious hangups on zaptel interface
Barry D. Hassler
- [asterisk-users] Are you using app_meetme or app_conference
Erick Perez
- [asterisk-users] UK Colocation services
mezzmor at aim.com
- [asterisk-users] SER with multiple asterisk deployment
Douglas Garstang
- [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P
Savoy, Kevin - Williston, ND
- [asterisk-users] How to detect dial tone on ZAP channel before
dialling using TDM2400P
Naija Man
- [asterisk-users] SMS Text Send working with BT Text in the UK??
Scott Stingel
- [asterisk-users] Are you using app_meetme or app_conference
Alyed Tzompa
- [asterisk-users] MWI on 1.4 Beta
Mark Hulber
- [asterisk-users] Spurious hangups on zaptel interface
Alyed Tzompa
- [asterisk-users] SIP on Asterisk, new install
joe, at j4computers
- [asterisk-users] Spurious hangups on zaptel interface
Alyed Tzompa
- [asterisk-users] SIP on Asterisk, new install
joe, at j4computers
- [asterisk-users] problem with trying to use two extensions for
different announcements
J. Duffy Beischel
- [asterisk-users] MOH problem
Andre Courchesne - Consultant
- [asterisk-users] Asterisk Hangups on PRI Interface
Sahil Gupta
- [asterisk-users] Asterisk Hangups on PRI Interface
Sahil Gupta
- [asterisk-users] SER with multiple asterisk deployment
Douglas Garstang
- [asterisk-users] Re: max number of devices in hint
Tomislav Parčina
- [asterisk-users] media stream count
Benjamin Jacob
- [asterisk-users] Re: Voip Buster - CID
Tomislav Parčina
- [asterisk-users] Re: ASTTAPI
Tomislav Parčina
- [asterisk-users] pay as you go t.t38 fax termination and origination
stan ford
- [asterisk-users] Re: RPID
Tomislav Parčina
- [asterisk-users] corrupt faxes
Louis-David Mitterrand
- [asterisk-users] Multiple asterisk same GUI
Sharon Lim
- RE: [asterisk-users] Re: ASTTAPI
Guido Hecken
- [asterisk-users] importance of crc4 in zaptel.conf?
Louis-David Mitterrand
- [asterisk-users] quadbri + tdm400p + modem-fax
Paco Brufal
- [asterisk-users] How does SIP work?
Vincent Delporte
- [asterisk-users] get the value from CDR
unplug
- [asterisk-users] Sangoma a301
adebayo omo-dare
- [asterisk-users] 407 Proxy Authentication Required
unplug
- [asterisk-users] No answer time
Dominik Kiełb
- [asterisk-users] 7940 vs. 7941
Julian Lyndon-Smith
- [asterisk-users] How to enable jingle in 1.4beta2?
Raffaele Porzio
- [asterisk-users] Is this phone any good?
Tim
- [asterisk-users] Asterisk <=> E1 <=> Alcatel OXO
Tomislav Parčina
- [asterisk-users] New ptlib dependency-requirement in SVN-trunk?
Brian Capouch
- [asterisk-users] X100M location in circuit requirement?
Nick Ellson
- [asterisk-users] Master.csv has stopped writing call logs.
Richard Reina
- [asterisk-users] asterisk on 2.4 kernel ... scheduler problem?
Cesc
- [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P
Ronnie Jones
- [asterisk-users] unable to call AT&T audio conference bridge
asterisk-user
- [asterisk-users] Contacts for Chan_gsm_bt maintainer?
Ed
- [asterisk-users] txfax reliability on TDM cards
Jerry Geis
- [asterisk-users] extensions.conf strangeness
Brian Candler
- SV: [asterisk-users] txfax reliability on TDM cards
Jon Schøpzinsky
- [asterisk-users] Polycom 501 One-way Audio
Paul Dugas
- [asterisk-users] WAS: 64 analog phones NOW: Selection criteri
a and recipie for a good Asterisk install [long]
Colin Anderson
- [asterisk-users] Is this phone any good?
Tim
- [asterisk-users] Asterisk -> Tekelec T6000 (Vocaldata, voiss)
Matthew Crocker
- [asterisk-users] Early rtp bridge and reINVITE in 1.4b2
H Quintana
- [asterisk-users] Re: Voip Buster - CID
Naija Man
- [asterisk-users] Polycom 501 One-way Audio
Bill Gibbs
- [asterisk-users] T1 incoming connects, but no sound
Nathan Bell
- [asterisk-users] Voicemail callback bug?
Kristian Kielhofner
- [asterisk-users] Polycom Queues, Login, Logout etc
Douglas Garstang
- [asterisk-users] Cisco CAll Manger and H323
Yusuf
- [asterisk-users] AstriCon 2006 Reminder / Hotel Selling Out
Steven Sokol
- [asterisk-users] Yet another processor question
Mark Farver
- [asterisk-users] RPID
Freddi Hansen
- [asterisk-users] You are not the next caller
Sean Cook
- [asterisk-users] Queue AddQueueMember()
Douglas Garstang
- [asterisk-users] Polycom Queues, Login, Logout etc
Douglas Garstang
- [asterisk-users] WAS: 64 analog phones NOW: Selection criteria
and recipie for a good Asterisk install [long]
Jeronimo Romero
- [asterisk-users] Asterisk lockup and boot error after one day?
Nick Ellson
- [Asterisk-Users] Help, please help -- IAX2 softphone to server
on LAN
Wolfgang_Borgon
- [asterisk-users] Forcing Transcode
Mr. Jones
- [asterisk-users] 7970G SIP8-0-4 not registering with asterisk
Richard Klingler
- [asterisk-users] Digium G.729 codec binaries updated for Asterisk
1.4 on Solaris
Jason Parker
- [asterisk-users] Forcing Transcode
Bill Gibbs
- [asterisk-users] Forcing Transcode
Colin Anderson
- [asterisk-users] SPA3000 and asterisk
Walter Willis
- [asterisk-users] Re: mISDN, 2 Billion HFC ISDN cards,
cannot dial or receive
Nigel Godfrey
- [asterisk-users] DUNDi Docs
Douglas Garstang
- [asterisk-users] t1-pri or sip trunk?
stan ford
- [asterisk-users] Cisco CAll Manger and H323
Dan Austin
- [asterisk-users] Is this phone any good
Tim
- [asterisk-users] pstn failback
stan ford
- [asterisk-users] Sirrix to Legacy PBX
Marnus van Niekerk
- [asterisk-users] Difference between "SIP Server" and "Outbound
Proxy"
Brian Candler
- [asterisk-users] re: asterisk/SER integration - HELP
Yair Hakak
- [asterisk-users] Asterisk -> Tekelec T6000 (Vocaldata, voiss)
Watkins, Bradley
- [Asterisk-Users] Off-hooking Snom hanset doesn't answer incoming call
Olivier
- [asterisk-users] VMware and Digium TDM400P card
Andy Green
- [asterisk-users] Replacing mpg123 with madplay under Solaris?
Frank Tarczynski
- [asterisk-users] Sip answer one side , ring other side
antonio
- [asterisk-users] Polycom IP430 HUM and Echo
Lorentz Hinrichsen
- [asterisk-users] attended transfer unreliable
Stefan Friedrich
- [asterisk-users] What minimum required packages for 1.4
news.gmane.org
- [asterisk-users] Extension Numbering
Norbert Zawodsky
- [asterisk-users] [Fwd: asterisk-users Digest, Vol 26, Issue 166]
asterisk-user
- [Asterisk-Users] Off-hooking Snom hanset doesn't answer incom
ing call
Colin Anderson
- [asterisk-users] Sip answer one side , ring other side
antonio
- [asterisk-users] t1-pri or sip trunk?
Colin Anderson
- [asterisk-users] real time billing system
Pato Valarezo
- [asterisk-users] Any Suggestions for Election Polling Application?
Gerald Drouillard
- [asterisk-users] SPA3000 register in asterisk
Walter Willis
- [asterisk-users] [Fwd: [Fwd: [Fwd: asterisk-users Digest, Vol 26,
Issue 166]]]
asterisk-user
- [asterisk-users] Asterisk IVR .wav issue
David Moring
- [asterisk-users] manager api redirect dropping calls
Wing Wong
- [asterisk-users] Cisco CAll Manger and H323
Dan Austin
- [asterisk-users] Queue and Pickup/DPickup
Ole Myhre
- [asterisk-users] recommended application for salesman using asterisk
Yu Safin
- [asterisk-users] Re: asterisk-users Digest, Vol 26, Issue 172
Frank Tarczynski
- [asterisk-users] Problems with DISA
Shidan
- [asterisk-users] A Step back with AddQueueMember() ?
Douglas Garstang
- [asterisk-users] SIP Gateway
Forrest Beck
- [asterisk-users] native sounds
Ed Nuñez
- [asterisk-users] Re: SIP Gateway
Forrest Beck
- [asterisk-users] Fiber Outage: LA->Washington
Jay R. Ashworth
- [asterisk-users] Queue AddQueueMember()
Douglas Garstang
- [asterisk-users] 480i phone: Is there a trick to registering with *
??
Colin Anderson
- [asterisk-users] FW: 480i phone: Is there a trick to registering
with * ?? <--forg ot my sip.conf
Colin Anderson
- [asterisk-users] SIP phones not talking
joe, at j4computers
- [asterisk-users] 480i phone: Is there a trick to registering with
*??
Gareth Owen
- [asterisk-users] Cisco CAll Manger and H323
Dan Austin
- [asterisk-users] 480i phone: Is there a trick to registering
with *??
Colin Anderson
- [asterisk-users] Realtime; User not registering {Urgent}
Abdul
- [asterisk-users] Queue Status via Dialplan
Michael J. Liberatore
- [asterisk-users] How do I reset a password?
Jim Lynch
- [asterisk-users] Re: Dropping extra frame of G.729 ?
Rosa De Santis
- [asterisk-users] Sip answer one side , ring other side
antonio
- [asterisk-users] Issues with Monitor in 1.4?
Ronald Lewis
- [asterisk-users] PRI channel becomming unavailable
Andre Courchesne - Consultant
- [asterisk-users] Asterisk Fax Server
Giedrius Augys
- [asterisk-users] Google talk and Asterisk 1.4
Robert LaPoint
- [asterisk-users] Where are the kernel sources?
Jim Lynch
- [asterisk-users] 480i phone: Is there a trick to registering
with *??
Colin Anderson
- [asterisk-users] Install issues on Freebsd 6.1 with Asterisk
1.4.0-beta2
boon dash
- [asterisk-users] Dial-9 (was Extension Numbering)
Naija Man
Last message date:
Sat Sep 30 20:54:35 MST 2006
Archived on: Sat Sep 30 20:54:37 MST 2006
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