Asterisk

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Elastix workshop in Toronto; Wed Nov 26th, 2008
- From: Simon P. Ditner
dial console/dsp hear crackling in headset
- From: Jerry Geis
Re: A question about how much an Asterisk Dcap consultant and a Sipmaster make
- From: Steve Totaro
A question about how much an Asterisk Dcap consultant and a Sipmaster make
- From: Scott Berry
Re: Using MAC or extension number as SIP identifier
- From: Philipp Kempgen
Re: SIP to IAX2 with delayed echo
- From: Drew Gibson
Rv: sas
- From: sas sas
VB 6 developer needed
- From: Gregory Malsack
Re: Using MAC or extension number as SIP identifier
- From: Eric \"ManxPower\" Wieling
Re: SIP to IAX2 with delayed echo
- From: Steve Totaro
Re: SIP to IAX2 with delayed echo
- From: Dave Fullerton
Re: How long will Asterisk 1.4.x supported/maintained
- From: Steve Totaro
Re: SIP to IAX2 with delayed echo
- From: Steve Totaro
Re: SIP to IAX2 with delayed echo
- From: c james
Re: SIP to IAX2 with delayed echo
- From: c james
Re: Using MAC or extension number as SIP identifier
- From: Olivier
Subversion Mirror Down for Maintenance
- From: Russell Bryant
Low RX volume and half duplex/"walkie-talkie" on AEX-804E
- From: Lincoln King-Cliby
Disable native bridge?
- From: Tod Fitch
Re: Using MAC or extension number as SIP identifier
- From: Eric \"ManxPower\" Wieling
Sending / Receiving sms messages with Portech 370
- From: Julian Lyndon-Smith
Re: Any other "free" toll free SIP providers out there?
- From: Tom Browning
Re: SIP to IAX2 with delayed echo
- From: Steve Totaro
Re: Using MAC or extension number as SIP identifier
- From: Daniel Hazelbaker
Re: Load balancing Asterisk.
- From: Alex Balashov
Re: SIP to IAX2 with delayed echo
- From: Tim Nelson
Re: two sip listening ports for single asterisk
- From: Matthew J. Roth
Re: setting up callback
- From: Mikhail (Plus Plus)
SIP to IAX2 with delayed echo
- From: c james
Re: Using MAC or extension number as SIP identifier
- From: Olivier
Re: Macro conversion in 1.6
- From: Joseph L. Casale
Re: Using MAC or extension number as SIP identifier
- From: Olivier
Re: Using MAC or extension number as SIP identifier
- From: Philipp Kempgen
Re: Using MAC or extension number as SIP identifier
- From: Steve Howes
Re: Using MAC or extension number as SIP identifier
- From: Daniel Hazelbaker
Re: Macro conversion in 1.6
- From: Atis Lezdins
Re: Load balancing Asterisk.
- From: Grey Man
Using MAC or extension number as SIP identifier
- From: Olivier
Re: Any other "free" toll free SIP providers out there?
- From: Atis Lezdins
Re: Load balancing Asterisk.
- From: Alex Balashov
Macro conversion in 1.6
- From: Joseph L. Casale
Re: Load balancing Asterisk.
- From: Nitzan Kon
Re: Role of asterisk
- From: Valentin Bud
Re: Load balancing Asterisk.
- From: Grygoriy Dobrovolskyy
Re: Load balancing Asterisk.
- From: SIP
jitterbuffer
- From: farah . auf
Re: Load balancing Asterisk.
- From: Alex Balashov
Re: Role of asterisk
- From: Bill Andersen
Re: Load balancing Asterisk.
- From: Alex Balashov
Re: Load balancing Asterisk.
- From: Nitzan Kon
Re: Load balancing Asterisk.
- From: Grey Man
Re: Load balancing Asterisk.
- From: Alex Balashov
Re: Load balancing Asterisk.
- From: Alex Balashov
Re: Load balancing Asterisk.
- From: SIP
Re: Load balancing Asterisk.
- From: Nitzan Kon
Re: puzzle
- From: Steve Totaro
Re: Load balancing Asterisk.
- From: Jeff LaCoursiere
Re: Collect digits from the Callee after the Call is connected.
- From: Doug Lytle
Re: Voicemail in Real Time
- From: Philipp Kempgen
Re: Load balancing Asterisk.
- From: Alex Balashov
Re: Load balancing Asterisk.
- From: Nitzan Kon
Re: Load balancing Asterisk.
- From: Grygoriy Dobrovolskyy
Load balancing Asterisk.
- From: Nitzan Kon
Voicemail in Real Time
- From: Ali Jawad
Re: Collect digits from the Callee after the Call is connected.
- From: Simith Nambiar
Re: Any other "free" toll free SIP providers out there?
- From: SIP
Re: Collect digits from the Callee after the Call is connected.
- From: Doug Lytle
Re: Collect digits from the Callee after the Call is connected.
- From: Darrin Henshaw
DTMF payload
- From: michel freiha
Collect digits from the Callee after the Call is connected.
- From: Simith Nambiar
Re: puzzle
- From: Philipp Kempgen
Re: Voice Mail
- From: Doug Lytle
Voice Mail
- From: michel freiha
Re: Configuring Sangoma BRI with zaptel?
- From: Claus Herwig
jitterbuffer
- From: farah . auf
Re: echo cancellation for sip phones
- From: Pezhman Lali
Re: Any other "free" toll free SIP providers out there?
- From: Tzafrir Cohen
Re: How long will Asterisk 1.4.x supported/maintained
- From: Tzafrir Cohen
Re: echo cancellation for sip phones
- From: Alan Lord
echo cancellation for sip phones
- From: Pezhman Lali
Re: How long will Asterisk 1.4.x supported/maintained
- From: Olivier
Re: HPEC performance
- From: Gordon Henderson
Re: dahdi_test drops after restarting Sangoma driver
- From: Hakan C
Re: two sip listening ports for single asterisk
- From: Mike
Re: puzzle
- From: Steve Totaro
Any other "free" toll free SIP providers out there?
- From: Tom Browning
Re: puzzle
- From: Jeff LaCoursiere
Re: puzzle
- From: Steve Totaro
Re: puzzle
- From: Alex Balashov
Re: puzzle
- From: Jeff LaCoursiere
Re: Upgrading Asterisk and FreePBX from 1.2 to 1.4
- From: Rob Hillis
Re: puzzle
- From: Steve Totaro
Re: puzzle
- From: Jeff LaCoursiere
Re: puzzle
- From: Alex Balashov
Re: Meetme "talker optimization" always on even when no "o" option present.
- From: Dan Austin
Re: puzzle
- From: Steve Totaro
Re: puzzle
- From: Jeff LaCoursiere
Re: puzzle
- From: Steve Totaro
dahdi_test drops after restarting Sangoma driver
- From: Andres
Re: HPEC performance
- From: Joseph L. Casale
Re: puzzle
- From: Jeff LaCoursiere
Re: HPEC performance
- From: Joseph L. Casale
Re: Asterisk 1.6 call files Disposition=NO ANSWER
- From: Steve Murphy
Re: puzzle
- From: Danny Nicholas
Re: puzzle
- From: Tzafrir Cohen
Re: puzzle
- From: Jeff LaCoursiere
VoiceMail - audio problem
- From: Shaun Wingrin
Re: Best way to handle include files?
- From: Doug
Re: puzzle
- From: Jeff LaCoursiere
Re: puzzle
- From: Tzafrir Cohen
Re: VoiceMail - audio problem
- From: David A. Bandel
Re: Asterisk 1.6 call files Disposition=NO ANSWER
- From: Steve Murphy
Re: HPEC performance
- From: Tzafrir Cohen
Re: Howto grab back call transfered from SIP phone
- From: Steve Totaro
Re: TDM400 (?) zap hangup
- From: Roderick A. Anderson
Re: Asterisk with or without OpenSER
- From: Alex Balashov
Re: Howto grab back call transfered from SIP phone
- From: Steve Totaro
Re: puzzle
- From: Danny Nicholas
Re: Asterisk with or without OpenSER
- From: mail-lists
Re: puzzle
- From: Jeff LaCoursiere
Re: puzzle
- From: Danny Nicholas
Re: puzzle
- From: Brent Davidson
Re: TDM400 (?) zap hangup
- From: Jared Smith
Re: HPEC performance
- From: Joseph L. Casale
Re: TDM400 (?) zap hangup
- From: Roderick A. Anderson
puzzle
- From: Jeff LaCoursiere
Re: Role of asterisk
- From: Valentin Bud
Re: Best way to handle include files?
- From: Tzafrir Cohen
Upgrading Asterisk and FreePBX from 1.2 to 1.4
- From: Carlos Chavez
Re: TDM400 (?) zap hangup
- From: Tzafrir Cohen
Best way to handle include files?
- From: Doug
TDM400 (?) zap hangup
- From: Roderick A. Anderson
Re: Howto grab back call transfered from SIP phone
- From: Chris Bagnall
Re: Asterisk with or without OpenSER
- From: Gordon Henderson
Re: question about connecting with Mobile Base Station
- From: John Todd
Re: VoiceMail - audio problem
- From: Shaun Wingrin
Re: Asterisk with or without OpenSER
- From: Alex Balashov
Re: Asterisk with or without OpenSER
- From: Alex Balashov
Re: Asterisk with or without OpenSER
- From: Steve Totaro
Re: Asterisk with or without OpenSER
- From: Gordon Henderson
Howto grab back call transfered from SIP phone
- From: Russell Brown
Re: Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working
- From: Brent Davidson
Re: IF else
- From: Atis Lezdins
Re: IF else
- From: Steve Edwards
Re: P2P
- From: Jim Dickenson
Re: Asterisk with or without OpenSER
- From: Alex Balashov
Re: Asterisk with or without OpenSER
- From: Steve Totaro
Re: Monitoring
- From: federico fetto
Re: help with dahdi
- From: Jerry Geis
Re: Asterisk with or without OpenSER
- From: mail-lists
Re: Monitoring
- From: Giorgio Ciccarelli
Re: Monitoring
- From: Jon Weisman
Re: help with dahdi
- From: Tzafrir Cohen
Re: IF else
- From: Atis Lezdins
Re: Monitoring
- From: Jon Weisman
Re: Role of asterisk
- From: Jared Smith
Re: help with dahdi
- From: Hakan C
Re: Asterisk NOW - Where to start - FOUND, Thanks
- From: Ferguson, Michael
Re: Role of asterisk
- From: Jared Smith
Re: IF else
- From: Gordon Henderson
Re: Monitoring
- From: federico fetto
Re: Monitoring
- From: Giorgio Ciccarelli
Asterisk NOW - Where to start
- From: Ferguson, Michael
Re: Monitoring
- From: Jon Weisman
Re: Monitoring
- From: Hakan C
Re: help with dahdi
- From: Tzafrir Cohen
Re: help with dahdi
- From: Jerry Geis
Re: P2P
- From: Valentin Bud
IF else
- From: michel freiha
Re: P2P
- From: ml@xxxxxxxxx
Re: Monitoring
- From: Jon Weisman
Re: asterisk conference
- From: Hakan C
Re: Picked up calls die in exactly 20 seconds
- From: Hakan C
Re: Configuring Sangoma BRI with zaptel?
- From: Hakan C
Re: Aeterisk NOW 1.5beta1 - CDR problem....
- From: Hakan C
Re: Monitoring
- From: Hakan C
Monitoring
- From: Jon Weisman
Re: P2P
- From: Valentin Bud
Re: question about connecting with Mobile Base Station
- From: Hakan C
Re: P2P
- From: Hakan C
Re: PoE switch recommendations?
- From: Mike Jagdis
P2P
- From: ml@xxxxxxxxx
Re: Forcing repacketization on SIP to SIP call
- From: Richard Brady
Role of asterisk
- From: Valentin Bud
Re: Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working
- From: Mikel Lindsaar
Re: help with dahdi
- From: Hakan C
Re: Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working
- From: Hakan C
Re: Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working
- From: Mikel Lindsaar
Re: help with dahdi
- From: Tzafrir Cohen
presence with polycom DND
- From: cfh
Re: question about connecting with Mobile Base Station
- From: mark morreny
Aeterisk NOW 1.5beta1 - CDR problem....
- From: Bipin
Re: two sip listening ports for single asterisk
- From: Matthew J. Roth
Re: help with dahdi
- From: Joseph L. Casale
Re: question about connecting with Mobile Base Station
- From: Andrew Joakimsen
question about connecting with Mobile Base Station
- From: mark morreny
Re: help with dahdi
- From: Jerry Geis
Re: help with dahdi
- From: Jerry Geis
Re: help with dahdi
- From: Joseph L. Casale
Re: Fwd: Polycom phone time behind one hour.
- From: Bryan M. Johns
help with dahdi
- From: Jerry Geis
Re: changing the size of voice packets
- From: Dan Austin
Asterisk 1.6 call files Disposition=NO ANSWER
- From: David Klaverstyn
Re: Caching Asterisk SIP useragent info?
- From: Paul Hales
Re: Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working
- From: Brent Davidson
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Steve Underwood
Re: Fwd: Polycom phone time behind one hour.
- From: Brent Davidson
Re: How to Barge specific extensions
- From: Hakan C
Fwd: Polycom phone time behind one hour.
- From: Doug Smith
Re: Do Digium Digital Cards Handle Remote Loopback Command?
- From: Kevin P. Fleming
Re: Incoming Transfer
- From: Godson Gera
Realtime MOH
- From: Sebastian
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Benny Amorsen
Re: Asterisk with or without OpenSER
- From: Yehavi Bourvine
Re: sound quality between two back-to-back asterisk
- From: Godson Gera
Re: Asterisk with or without OpenSER
- From: John Todd
Re: Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working
- From: Mikel Lindsaar
setting up callback
- From: Михаил (Плюс Плюс)
Re: Do Digium Digital Cards Handle Remote Loopback Command?
- From: Alex Balashov
Do Digium Digital Cards Handle Remote Loopback Command?
- From: George Pajari
Re: diax debian package
- From: Tzafrir Cohen
diax debian package
- From: Joseph
Re: Asterisk 1.4.21.2 and gtalk2voip
- From: Administrator TOOTAI
sound quality between two back-to-back asterisk
- From: mark morreny
Re: meetme command from 1.4 to 1.6
- From: Jeff Peeler
Re: FOP with Asterisk 1.6. No call Information.
- From: Carlos Chavez
Incoming Transfer
- From: Joseph L. Casale
Configuring Sangoma BRI with zaptel?
- From: Claus Herwig
Re: Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working
- From: Mikel Lindsaar
Re: Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working
- From: Tony Mountifield
Re: How to Barge specific extensions
- From: troxlinux
Crash when rebooting or unload xorcom modules
- From: Loic Didelot
Asterisk 1.4.21.2 and gtalk2voip
- From: Administrator TOOTAI
Re: FOP with Asterisk 1.6. No call Information.
- From: Doug Lytle
Re: Asterisk with or without OpenSER
- From: Steve Totaro
meetme command from 1.4 to 1.6
- From: Jerry Geis
Re: Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working
- From: Mikel Lindsaar
Re: Picked up calls die in exactly 20 seconds
- From: Steve Totaro
Re: Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working
- From: Mikel Lindsaar
Asterisk with or without OpenSER
- From: Yehavi Bourvine
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Steve Underwood
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Tony Mountifield
FOP with Asterisk 1.6. No call Information.
- From: David Klaverstyn
Re: busy-level / busy-limit Asterisk 1.4.22
- From: nik600
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Benny Amorsen
Re: busy-level / busy-limit Asterisk 1.4.22
- From: Steve Howes
busy-level / busy-limit Asterisk 1.4.22
- From: nik600
Re: HPEC performance
- From: Gordon Henderson
Re: Caching Asterisk SIP useragent info?
- From: Veselin K
Asterisk Realtime and device contexts
- From: Sawan Vithlani
Re: HPEC performance
- From: Tzafrir Cohen
Re: test OpenVox B400P and junghans card for dahdi BRI wcb4xxp
- From: Tzafrir Cohen
Re: How long will Asterisk 1.4.x supported/maintained
- From: Tzafrir Cohen
Re: PoE switch recommendations?
- From: Jesse Molina
test OpenVox B400P and junghans card for dahdi BRI wcb4xxp
- From: lizhong zhu
How to Barge specific extensions
- From: amit salunkhe
Re: MixMonitor Problem
- From: Trevor Peirce
Re: Polycom low volume
- From: Doug
HPEC performance
- From: Joseph L. Casale
Re: How long will Asterisk 1.4.x supported/maintained
- From: Tilghman Lesher
Re: How long will Asterisk 1.4.x supported/maintained
- From: Atis Lezdins
Re: two sip listening ports for single asterisk
- From: Robert McNaught
Re: Deny FOP originated calls
- From: Tilghman Lesher
Message 12216
- From: asterisk-users
two sip listening ports for single asterisk
- From: Rizwan Hisham
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Matthew Fredrickson
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Matthew Fredrickson
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Steve Underwood
Re: upgrade to 1.6
- From: Matt Riddell
Re: ALL of DIDx Down?
- From: Matt Riddell
Re: Caching Asterisk SIP useragent info?
- From: Paul Hales
Re: How long will Asterisk 1.4.x supported/maintained
- From: Philipp Kempgen
Picked up calls die in exactly 20 seconds
- From: Juan Carlos Castro y Castro
Re: How long will Asterisk 1.4.x supported/maintained
- From: Tilghman Lesher
Re: How long will Asterisk 1.4.x supported/maintained
- From: Anthony Francis
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Wilton Helm
Re: How long will Asterisk 1.4.x supported/maintained
- From: Tilghman Lesher
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Matthew Fredrickson
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Benny Amorsen
Re: Polycom low volume
- From: hin lee
Re: Digium Card Noice issue
- From: Benny Amorsen
Re: Full Duplex
- From: Matthew Fredrickson
MixMonitor Problem
- From: Sebastian
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Matthew Fredrickson
Re: Digium Card Noice issue
- From: Matthew Fredrickson
Re: How long will Asterisk 1.4.x supported/maintained
- From: Philipp Kempgen
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Wilton Helm
Re: Full Duplex
- From: Singer X.J. Wang
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: John Todd
Re: Full Duplex
- From: Doug Lytle
Re: Digium Card Noice issue
- From: Andrew Joakimsen
Re: Full Duplex
- From: Ken Williams
Deny FOP originated calls
- From: Rodolfo Alcazar Portillo
Re: Full Duplex
- From: Ken Williams
Re: Full Duplex
- From: Doug Lytle
Re: Full Duplex
- From: Gordon Henderson
Re: AMI Events disabling.
- From: Simith Nambiar
Re: Hints and realtime
- From: Tilghman Lesher
Re: Digium Card Noice issue
- From: Steve Howes
Re: Caching Asterisk SIP useragent info?
- From: Tilghman Lesher
dahdi and asterisk 1.4.22
- From: Jerry Geis
Re: Debugging Asterisk
- From: Tilghman Lesher
Digium Card Noice issue
- From: Bipin
Hints and realtime
- From: Julian Lyndon-Smith
AMI Events disabling.
- From: Simith Nambiar
Re: no dial to busy sip line
- From: Christophorus Laube
Re: Debugging Asterisk
- From: Atis Lezdins
Full Duplex
- From: Ken Williams
Re: Debugging Asterisk
- From: Terry Wilson
asterisk conference
- From: Giedrius Augys
ALL of DIDx Down?
- From: Andrew Joakimsen
Re: upgrade to 1.6
- From: Jerry Geis
E1/channels
- From: Khaled Chehab
Re: upgrade to 1.6
- From: Hakan C
Re: Queue App - Set monitoring dynamically
- From: equis software
Re: Caching Asterisk SIP useragent info?
- From: andrea
Re: Caching Asterisk SIP useragent info?
- From: Veselin K
Re: Debugging Asterisk
- From: Valentin Bud
Debugging Asterisk
- From: Mikel Lindsaar
Re: IAX2 client for 'eee pc 1000'
- From: Roger Marquis
upgrade to 1.6
- From: Jerry Geis
Re: dahdi compile error on svn
- From: Jerry Geis
Re: dahdi compile error on svn
- From: Jerry Geis
Re: Caching Asterisk SIP useragent info?
- From: Paul Hales
Re: dahdi compile error on svn
- From: Tzafrir Cohen
Record Application
- From: Kannan Murali
Re: PBX -> PRI -> * -> Telco not working
- From: Tony Mountifield
Re: Queue App - Set monitoring dynamically
- From: Matt Riddell
Re: iPhone SIP or IAX client (without proxy)?
- From: Eric Chamberlain
dahdi compile error on svn
- From: Jerry Geis
DialPlan
- From: michel freiha
Caching Asterisk SIP useragent info?
- From: veselin
Re: IAX2 client for "eee pc 1000"
- From: Tzafrir Cohen
iPhone SIP or IAX client (without proxy)?
- From: Lincoln King-Cliby
Re: IAX2 client for "eee pc 1000"
- From: Joseph
* + Legacy PBX works but strange problem
- From: Sriram
Re: What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux software)?
- From: Bill Michaelson
Re: * + Legacy PBX works but strange problem
- From: ram
* + Legacy PBX works but strange problem
- From: Sriram
Re: IAX2 client for "eee pc 1000"
- From: henry
Re: Polycom low volume
- From: Michael Graves
Re: * + Legacy PBX works but strange problem
- From: Steve Totaro
Info about dstchannel
- From: Chris Maciejewski
Re: * + Legacy PBX works but strange problem
- From: Robert Boardman
* + Legacy PBX works but strange problem
- From: Sriram
Re: IAX2 client for "eee pc 1000"
- From: Tzafrir Cohen
Re: Polycom low volume
- From: hin lee
Re: IAX2 client for "eee pc 1000"
- From: Joseph
Re: IAX2 client for "eee pc 1000"
- From: Rob Hillis
Re: Polycom low volume
- From: Doug
Re: Polycom low volume
- From: hin lee
Re: IAX2 client for "eee pc 1000"
- From: Alex Balashov
Re: IAX2 client for "eee pc 1000"
- From: Alex Robar
Re: PBX -> PRI -> * -> Telco not working
- From: Mikel Lindsaar
Re: IAX2 client for "eee pc 1000"
- From: Joseph
Re: IAX2 client for "eee pc 1000"
- From: Alex Robar
Re: IAX2 client for "eee pc 1000"
- From: David
Re: IAX2 client for "eee pc 1000"
- From: Joseph
Re: Asterisk GUI and SIP registration
- From: Brandon Kruse
Re: IAX2 client for "eee pc 1000"
- From: David
RV: MixMonitor and Queues
- From: Sebastian
IAX2 client for "eee pc 1000"
- From: Joseph
Re: PBX -> PRI -> * -> Telco not working
- From: Tony Mountifield
Asterisk GUI and SIP registration
- From: Joseph L. Casale
Re: Polycom low volume
- From: Darrick Hartman
Re: PBX -> PRI -> * -> Telco not working
- From: Mikel Lindsaar
MixMonitor and Queues
- From: Sebastian
Re: Polycom low volume
- From: Michael Graves
Re: PBX -> PRI -> * -> Telco not working
- From: Mikel Lindsaar
Re: PBX -> PRI -> * -> Telco not working
- From: Mikel Lindsaar
Re: PBX -> PRI -> * -> Telco not working
- From: Mikel Lindsaar
Re: PBX -> PRI -> * -> Telco not working
- From: Tony Mountifield
Polycom low volume
- From: hin lee
Re: RTP LOG
- From: Atis Lezdins
Re: Looking for a good lightweight Linux softPhone
- From: Atis Lezdins
Re: music on hold
- From: fateme fatah
Re: RTP LOG
- From: Max Alex
PBX -> PRI -> * -> Telco not working
- From: Mikel Lindsaar
Re: Preserving DID numbers on PRI pass through
- From: Mikel Lindsaar
Re: AS5200 <-> T100P - No alarms but no calls either...
- From: Don Fanning
Best way to handle include files?
- From: Doug
Re: installation
- From: Steve Totaro
installation
- From: DesRae Mason
Re: RTP LOG
- From: Benny Amorsen
Re: asterisk/E1
- From: Valentin Bud
Re: asterisk/E1
- From: Edwin Lam
Re: Looking for a good lightweight Linux softPhone
- From: Tzafrir Cohen
Linksys SPA 400, 901 and 921 with asterisk
- From: Valentin Bud
Originate on AMI
- From: Marco Eduardo Cordeiro
Manilla inbound DID
- From: Fred Posner
Re: Looking for a good lightweight Linux softPhone
- From: Atis Lezdins
PRI users, please read
- From: Tilghman Lesher
Re: Looking for a good lightweight Linux softPhone
- From: zoachien@xxxxxxxxxxx
Re: Looking for a good lightweight Linux softPhone
- From: Gordon Henderson
Re: Looking for a good lightweight Linux softPhone
- From: Jeff LaCoursiere
Re: Looking for a good lightweight Linux softPhone
- From: Gordon Henderson
Re: Looking for a good lightweight Linux softPhone
- From: Tilghman Lesher
Re: no dial to busy sip line
- From: David Gibbons
no dial to busy sip line
- From: Christophorus Laube
Re: Looking for a good lightweight Linux softPhone
- From: Tzafrir Cohen
Re: Looking for a good lightweight Linux softPhone
- From: Gordon Henderson
Queue App - Set monitoring dynamically
- From: equis software
Re: Looking for a good lightweight Linux softPhone
- From: Tzafrir Cohen
Re: asterisk/E1
- From: Steve Totaro
Re: Looking for a good lightweight Linux softPhone
- From: Alan Lord
Re: RTP LOG
- From: Positively Optimistic
Re: asterisk/E1
- From: Tony Mountifield
Re: kick from conference message on 1.2.23
- From: Tony Mountifield
Re: asterisk/E1
- From: Khaled Chehab
Re: Looking for a good lightweight Linux softPhone
- From: Cory Andrews
Re: asterisk/E1
- From: Steve Totaro
Re: Looking for a good lightweight Linux softPhone
- From: Alex Balashov
Looking for a good lightweight Linux softPhone
- From: Gordon Henderson
Re: asterisk/E1
- From: Khaled Chehab
Re: asterisk/E1
- From: Godson Gera
Re: asterisk/E1
- From: Steve Totaro
kick from conference message on 1.2.23
- From: Jerry Geis
asterisk/E1
- From: Khaled Chehab
Re: ParkandAnnounce?
- From: Doug Lytle
Re: Preserving DID numbers on PRI pass through
- From: Tony Mountifield
Re: Dedicated Servers
- From: Tom Moore
RTP LOG
- From: Max Alex
Dedicated Servers
- From: Sahil Gupta
openLDAP
- From: jude NOA
Virtual Question
- From: Babcock, Michael Alex
ParkandAnnounce?
- From: Positively Optimistic
Preserving DID numbers on PRI pass through
- From: Mikel Lindsaar
Re: Voicemail IMAP ./configure error
- From: Mark Michelson
Re: Asterisk and Zaptel version numbers -- how close is close enough?
- From: Steve Edwards
Asterisk and Zaptel version numbers -- how close is close enough?
- From: Steve Edwards
Re: Why Nat=yes Nat=no Option?
- From: Alex Balashov
Re: Voicemail IMAP ./configure error
- From: c james
Re: Why Nat=yes Nat=no Option?
- From: Klaus Darilion
Re: database queries from extensions.conf
- From: Wolfgang Pichler
Re: Why Nat=yes Nat=no Option?
- From: Klaus Darilion
Re: database queries from extensions.conf
- From: Klaus Darilion
Re: database queries from extensions.conf
- From: Klaus Darilion
Re: How long will Asterisk 1.4.x supported/maintained
- From: Tilghman Lesher
Re: Why Nat=yes Nat=no Option?
- From: Alex Balashov
Re: Why Nat=yes Nat=no Option?
- From: Alex Balashov
Re: Why Nat=yes Nat=no Option?
- From: Klaus Darilion
Re: Why Nat=yes Nat=no Option?
- From: Klaus Darilion
Re: Why Nat=yes Nat=no Option?
- From: Alex Balashov
Re: Why Nat=yes Nat=no Option?
- From: Steve Totaro
Re: Why Nat=yes Nat=no Option?
- From: Alex Balashov
Re: Why Nat=yes Nat=no Option?
- From: Steve Totaro
Re: database queries from extensions.conf
- From: Wolfgang Pichler
Re: 1.6 Production ready??
- From: César García
Re: database queries from extensions.conf
- From: Jared Smith
Re: Why Nat=yes Nat=no Option?
- From: Alex Balashov
Re: Why Nat=yes Nat=no Option?
- From: Alex Balashov
Re: asterisk setup w/ voIP phones
- From: Justin Case
Re: Why Nat=yes Nat=no Option?
- From: Klaus Darilion
database queries from extensions.conf
- From: Klaus Darilion
How long will Asterisk 1.4.x supported/maintained
- From: Klaus Darilion
[Fwd: [OpenSIPS-Devel] RFC: new opensips design]
- From: Bogdan-Andrei Iancu
Tos_sip
- From: Anael DIAZ
Re: 1.6 Production ready??
- From: Sherwood McGowan
Problems with Licensed g729a codec from Digium
- From: Murray Blakeman
Re: Parking help - causing Asterisk crash
- From: Doug Lytle
cisco voice gw / cisco call manager /asterisk for voice record, ivr
- From: Litzler Mihály
Re: asterisk setup w/ voIP phones
- From: Gordon Henderson
Parking help - causing Asterisk crash
- From: Daniel Johnson
Re: asterisk setup w/ voIP phones
- From: Matt Gibson
Re: How to get correct dial result for outgoing calls thru ISDN?
- From: Tilghman Lesher
asterisk setup w/ voIP phones
- From: Mike
Re: One little pill and you are on your way to sex!
- From: asterisk-users
Re: How to get correct dial result for outgoing calls thru ISDN?
- From: Eric \"ManxPower\" Wieling
Re: E1 PRI to and from SIP screeching
- From: Peter Lindquist
Re: AsteriskNOW 1.5 - app_voicemail_imapstorage.so won't talk to IMAP server
- From: Jason Lixfeld
Re: How to get correct dial result for outgoing calls thru ISDN?
- From: Tilghman Lesher
Re: E1 PRI to and from SIP screeching
- From: Eric \"ManxPower\" Wieling
Re: How to get correct dial result for outgoing calls thru ISDN?
- From: Eric \"ManxPower\" Wieling
Re: Why Nat=yes Nat=no Option?
- From: Eric \"ManxPower\" Wieling
Re: E1 PRI to and from SIP screeching
- From: Peter Lindquist
test OpenVox B400P and junghans card for dahdi BRI wcb4xxp
- From: lizhong zhu
Re: What makes TDM400 FXS Connection to TELCO go into Off Hook State?
- From: Tzafrir Cohen
Re: set(CALLERID(name) not working
- From: sean darcy
Re: Why Nat=yes Nat=no Option?
- From: Steve Totaro
Re: What are the minimum realtime fields for sipusers?
- From: Eric Chamberlain
Re: Why Nat=yes Nat=no Option?
- From: Tilghman Lesher
Re: Why Nat=yes Nat=no Option?
- From: Steve Totaro
Re: What makes TDM400 FXS Connection to TELCO go into Off Hook State?
- From: Jim Duda
1.4.22 CALLERID(num)
- From: James Fromm
Re: Why Nat=yes Nat=no Option?
- From: Alex Balashov
Re: CANCEL FORWAR
- From: Alex Balashov
Re: CANCEL FORWAR
- From: Alex Balashov
CANCEL FORWAR
- From: Bruno Rodrigues
Re: List eating mail again?
- From: Atis Lezdins
Re: List eating mail again?
- From: Doug Lytle
Re: Use DECT GAP handsets with Snom M3 base?
- From: Paul Chambers
Re: QueueLog from AMI
- From: Atis Lezdins
List eating mail again?
- From: Doug Lytle
Re: set(CALLERID(name) not working
- From: sean darcy
Re: Why Nat=yes Nat=no Option?
- From: Steve Totaro
Re: set(CALLERID(name) not working
- From: Matt Riddell
Re: Why Nat=yes Nat=no Option?
- From: Alex Balashov
Re: Why Nat=yes Nat=no Option?
- From: Steve Totaro
Re: set(CALLERID(name) not working
- From: Doug Lytle
Re: Why Nat=yes Nat=no Option?
- From: Alex Balashov
Re: SIP provider and NAT
- From: Sebastian
Why Nat=yes Nat=no Option?
- From: Steve Totaro
Re: Voicemail IMAP ./configure error
- From: Mark Michelson
Re: set(CALLERID(name) not working
- From: sean darcy
Re: SIP provider and NAT
- From: Steve Totaro
Re: SIP provider and NAT
- From: Alex Balashov
Re: 1.4.22 vs 1.4.21.2 - IAX2 regression ?
- From: Igor Zamocky
SIP provider and NAT
- From: Jim Dickenson
Re: Request for testing of new driver for B410P Quad-Port BRI
- From: stoffell
Re: PSTN Channels merging with SIP channels!!!
- From: R_O_L_A_N_D
Re: AsteriskNOW 1.5 - app_voicemail_imapstorage.so won't talk to IMAP server
- From: Jason Parker
Re: QueueLog from AMI
- From: Sebastian Gutierrez
Re: QueueLog from AMI
- From: Atis Lezdins
Re: PSTN Channels merging with SIP channels!!!
- From: Dave Fullerton
How to get correct dial result for outgoing calls thru ISDN?
- From: Daniel - Asterisk
Re: E1 PRI to and from SIP screeching
- From: Steve Totaro
PSTN Channels merging with SIP channels!!!
- From: R_O_L_A_N_D
Re: QueueLog from AMI
- From: Sebastian Gutierrez
E1 PRI to and from SIP screeching
- From: Peter Lindquist
Re: QueueLog from AMI
- From: Atis Lezdins
What are the minimum realtime fields for sipusers?
- From: Eric Chamberlain
QueueLog from AMI
- From: Sebastian Gutierrez
Query about Call Recording with Asterisk / Freeswitch in Cisco IPCC deployment
- From: Kashif Naeem
Re: test OpenVox B400P and junghans card for dahdi BRI wcb4xxp
- From: Tzafrir Cohen
Re: AS5200 <-> T100P - No alarms but no calls either...
- From: Steve Howes
Re: ztdummy: rtc: lost some interrupts at 1024Hz
- From: Giorgio Incantalupo
test OpenVox B400P and junghans card for dahdi BRI wcb4xxp
- From: lizhong zhu
The sound is played but I did not hear
- From: jhon digital21
Re: Grandstream and pickup
- From: Julian Lyndon-Smith
Re: Grandstream and pickup
- From: Gordon Henderson
Re: Grandstream and pickup
- From: Julian Lyndon-Smith
Re: play file from url
- From: Pezhman Lali
Re: AS5200 <-> T100P - No alarms but no calls either...
- From: Tzafrir Cohen
Re: What makes TDM400 FXS Connection to TELCO go into Off Hook State?
- From: Tzafrir Cohen
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Wilton Helm
Re: set(CALLERID(name) not working
- From: Daniel Lynes
Re: set(CALLERID(name) not working
- From: sean darcy
Re: set(CALLERID(name) not working
- From: sean darcy
Re: Use DECT GAP handsets with Snom M3 base?
- From: Michael Graves
Use DECT GAP handsets with Snom M3 base?
- From: Paul Chambers
Re: DNS A queries for channel
- From: Matt Riddell
AS5200 <-> T100P - No alarms but no calls either...
- From: Don Fanning
Re: Grandstream and pickup
- From: Julian Lyndon-Smith
Re: OT: Polycom Firmware available (by accident?)
- From: Darryl Dunkin
Grandstream and pickup
- From: Julian Lyndon-Smith
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Steve Underwood
Re: What makes TDM400 FXS Connection to TELCO go into Off Hook State?
- From: Jim Duda
Re: AsteriskNOW 1.5 - app_voicemail_imapstorage.so won't talk to IMAP server
- From: Jason Parker
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Steve Murphy
AsteriskNOW 1.5 - app_voicemail_imapstorage.so won't talk to IMAP server
- From: Jason Lixfeld
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Wilton Helm
Request for testing of new driver for B410P Quad-Port BRI
- From: Shaun Ruffell
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Steve Murphy
Re: play file from url
- From: Singer X.J. Wang
play file from url
- From: Mike Clark
Re: Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Steve Totaro
Use the NEW ulaw/alaw codecs (slower, but cleaner)
- From: Karl Fife
Re: view the current calls and their codec
- From: Anthony Francis
Re: ztdummy: rtc: lost some interrupts at 1024Hz
- From: JR Richardson
Re: OT: Polycom Firmware available (by accident?)
- From: Doug
Re: view the current calls and their codec
- From: nik600
Re: ztdummy: rtc: lost some interrupts at 1024Hz.
- From: Remi Quezada
Re: OT: Polycom Firmware available (by accident?)
- From: Dave Fullerton
ztdummy: rtc: lost some interrupts at 1024Hz.
- From: Giorgio Incantalupo
Re: OT: Polycom Firmware available (by accident?)
- From: Tilghman Lesher
Re: DNS A queries for channel
- From: Tilghman Lesher
Re: GEN-GEN and Manual Ring-Down (MRD)?
- From: Jorge Mendoza
Re: view the current calls and their codec
- From: Tim Nelson
Re: Server for 25-30 phones, sip trunks over the net
- From: Ben Hauger
Re: view the current calls and their codec
- From: nik600
Re: OT: Polycom Firmware available (by accident?)
- From: Dave Fullerton
Re: OT: Polycom Firmware available (by accident?)
- From: Jared Smith
Re: OT: Polycom Firmware available (by accident?)
- From: Doug Lytle
Re: OT: Polycom Firmware available (by accident?)
- From: Vinícius Fontes
Asterisk CDR Error ??
- From: Ruddy Gbaguidi
music on hold
- From: Uros Djokic
Re: DNS A queries for channel
- From: samuel
OT: Polycom Firmware available (by accident?)
- From: Dave Fullerton
Re: view the current calls and their codec
- From: Tim Nelson
Re: Dial outside number using the E1 Link
- From: Pezhman Lali
view the current calls and their codec
- From: nik600
Re: music on hold
- From: Peter Evans
Re: TE410P alarms stay RED with 1.4.22
- From: Louis-David Mitterrand
Re: Voicemail IMAP ./configure error
- From: c james
help with call with no sound via PSTN
- From: César García
Re: changing the size of voice packets
- From: Pezhman Lali
Re: music on hold
- From: Jeff LaCoursiere
Re: DNS A queries for channel
- From: Philipp Kempgen
Re: Inbound/Outbound undesired behavior
- From: Lenz Emilitri
Re: TE410P alarms stay RED with 1.4.22
- From: Laurent Caron
Dial outside number using the E1 Link
- From: fateme fatah
Re: TE410P alarms stay RED with 1.4.22
- From: Laurent Caron
Re: Forcing repacketization on SIP to SIP call
- From: Richard Brady
Re: DNS A queries for channel
- From: Matt Riddell
Re: music on hold
- From: 邱磊
Re: music on hold
- From: 邱磊
Re: DNS A queries for channel
- From: samuel
Re: What makes TDM400 FXS Connection to TELCO go into Off Hook State?
- From: Tzafrir Cohen
TE410P alarms stay RED with 1.4.22
- From: Louis-David Mitterrand
Re: music on hold
- From: Tzafrir Cohen
Re: OT: Disable Polycom 650 Forward Softkey
- From: Andrew Joakimsen
Re: GEN-GEN and Manual Ring-Down (MRD)?
- From: Raj Jain
Re: music on hold
- From: Peter Evans
Re: music on hold
- From: Lee, John (Sydney)
music on hold
- From: 邱磊
Re: Server for 25-30 phones, sip trunks over the net
- From: Gordon Henderson
dial a number while play the sound
- From: 邱磊
Re: set(CALLERID(name) not working
- From: C F
Server for 25-30 phones, sip trunks over the net
- From: nb
What makes TDM400 FXS Connection to TELCO go into Off Hook State?
- From: Jim Duda
Re: Help with asterisk and avaya SIP trunking
- From: Shaun Ewing
Re: Voicemail IMAP ./configure error
- From: Mark Michelson
Re: changing the size of voice packets
- From: Kristian Kielhofner
Re: changing the size of voice packets
- From: Kevin P. Fleming
Re: GEN-GEN and Manual Ring-Down (MRD)?
- From: Tilghman Lesher
Re: Voicemail IMAP ./configure error
- From: c james
Re: changing the size of voice packets
- From: Kristian Kielhofner
Re: changing the size of voice packets
- From: Alex Balashov
Re: changing the size of voice packets
- From: Kevin P. Fleming
Re: DNS A queries for channel
- From: Matt Riddell
Re: changing the size of voice packets
- From: Alex Balashov
Re: Voicemail IMAP ./configure error
- From: Mark Michelson
Re: directrtpsetup without reinvite
- From: regs@xxxxxxxxxx
Voicemail IMAP ./configure error
- From: c james
Re: Asterisk daemon dies about once per day
- From: Steve Murphy
Re: changing the size of voice packets
- From: Kristian Kielhofner
Re: changing the size of voice packets
- From: Kristian Kielhofner
Re: changing the size of voice packets
- From: Alex Balashov
Fwd: console/dsp asterisk seg fault
- From: Steve Totaro
Re: changing the size of voice packets
- From: John Todd
Re: console/dsp asterisk seg fault
- From: Jerry Geis
Re: console/dsp asterisk seg fault
- From: Jerry Geis
Re: Asterisk daemon dies about once per day
- From: Steve Murphy
Re: Asterisk daemon dies about once per day
- From: Tim Nelson
Re: Help with asterisk and avaya SIP trunking
- From: Krishna Sumanth Chava
Re: Recommend Wireless IP Phone
- From: Wilton Helm
Asterisk daemon dies about once per day
- From: Douglas Mortensen
Re: console/dsp asterisk seg fault
- From: Steve Totaro
console/dsp asterisk seg fault
- From: Jerry Geis
Re: directrtpsetup without reinvite
- From: Kevin P. Fleming
Re: directrtpsetup without reinvite
- From: Alex Balashov
Re: directrtpsetup without reinvite
- From: Kristian Kielhofner
Re: Recommend Wireless IP Phone
- From: Michael Graves
Re: Using AMI to determine PRI Channels Used
- From: Anthony Francis
Re: Using AMI to determine PRI Channels Used
- From: Richard Lyman
Re: Recommend Wireless IP Phone
- From: Eric \"ManxPower\" Wieling
Re: Codec problems when using G.723
- From: Thomas Winter
Re: Recommend Wireless IP Phone
- From: Drew Gibson
Re: Sendmail using SMTP authorization
- From: hugolivude
Re: Codec problems when using G.723
- From: Eric \"ManxPower\" Wieling
Re: Codec problems when using G.723
- From: Thomas Winter
SRTP support in asterisk 1.6
- From: Max Alex
Re: Using AMI to determine PRI Channels Used
- From: Godson Gera
Using AMI to determine PRI Channels Used
- From: David Budny
Re: TE121B Doesn't Fit PCI-E Slot
- From: David Budny
directrtpsetup without reinvite
- From: regs@xxxxxxxxxx
Re: changing the size of voice packets
- From: Pezhman Lali
Re: DNS A queries for channel
- From: samuel
Re: set(CALLERID(name) not working
- From: Doug Lytle
analog issues using xen virtualization
- From: Jesus Lee
GEN-GEN and Manual Ring-Down (MRD)?
- From: Tony Mountifield
Re: changing the size of voice packets
- From: Alex Balashov
Re: DNS A queries for channel
- From: samuel
Re: changing the size of voice packets
- From: Igor Goncharovsky
changing the size of voice packets
- From: Pezhman Lali
Re: crashes after upgrade from 1.2.16 to 1.4.21.2
- From: Louis-David Mitterrand