July 2003 Archives by author
Starting: Tue Jul 1 01:12:16 MST 2003
Ending: Thu Jul 31 22:28:56 MST 2003
Messages: 2198
- [Asterisk-Users] E1 R2 on Asterisk
LQ (Asterisk)
- [Asterisk-Users] E1 R2 on Asterisk
LQ (Asterisk)
- [Asterisk-Users] E1 R2 on Asterisk
LQ (Asterisk)
- [Asterisk-Users] Asterisk on Cygwin?
Chris Earle (CBL)
- [Asterisk-Users] ATA-186 software upgrade 2.16.1 - notes?
Chris Earle (CBL)
- [Asterisk-Users] OT: list format vs newsgroup format
Chris Earle (CBL)
- [Asterisk-Users] OT: list format vs newsgroup format
Chris Earle (CBL)
- [Asterisk-Users] OT: list format vs newsgroup format
Chris Earle (CBL)
- [Asterisk-Users] OT: list format vs newsgroup format
Chris Earle (CBL)
- [Asterisk-Users] Asterix Manual
Dhammika Gunawardena (ISP)
- [Asterisk-Users] Asterix Manual
Dhammika Gunawardena (ISP)
- [Asterisk-Users] Asterisk and Hot Desks??
WipeOut .
- [Asterisk-Users] Asterisk and Hot Desks??
WipeOut .
- [Asterisk-Users] Seg Fault!!
WipeOut .
- [Asterisk-Users] Seg Fault!!
WipeOut .
- [Asterisk-Users] Bugetone NTP problem..
WipeOut .
- [Asterisk-Users] Bugetone SIP Transfer
WipeOut .
- [Asterisk-Users] Is there any real asterisk documentation ?
WipeOut .
- [Asterisk-Users] How does Asterisk handle connecting two IP
end points?
WipeOut .
- [Asterisk-Users] How do you force Asterisk to use only
specific codecs?
WipeOut .
- [Asterisk-Users] How do you force Asterisk to use only
specific codecs?
WipeOut .
- [Asterisk-Users] Need a recommendation on a good
motherboard/processor combination
WipeOut .
- [Asterisk-Users] That is not a valid conference number meesage
WipeOut .
- [Asterisk-Users] switch => priority in the dialplan.. (probably an issue for Mark)
WipeOut .
- [Asterisk-Users] switch => priority in the dialplan..
(probably an issue for Mark)
WipeOut .
- [Asterisk-Users] switch => priority in the dialplan..
(probably an issue for Mark)
WipeOut .
- [Asterisk-Users] Voicemail2 Contexts
WipeOut .
- [Asterisk-Users] IAX Bandwidth Question
WipeOut .
- [Asterisk-Users] Remote * Using IAX
WipeOut .
- [Asterisk-Users] Asterisk and VMWare
WipeOut .
- [Asterisk-Users] System command..
WipeOut .
- [Asterisk-Users] SIp Registration
WipeOut .
- [Asterisk-Users] Switch issues with non-dedicated comms.. (My experience)
WipeOut .
- [Asterisk-Users] Budgetone and Voicemail
WipeOut .
- [Asterisk-Users] Chan_capi hanging channels
WipeOut .
- [Asterisk-Users] Music on hold quality..
WipeOut .
- [Asterisk-Users] Music on hold quality..
WipeOut .
- [Asterisk-Users] BudgeTone-100 Date and Time
WipeOut .
- [Asterisk-Users] wait and user input..
WipeOut .
- [Asterisk-Users] wait and user input..
WipeOut .
- [Asterisk-Users] What does "callerid=" in sip.conf do?
WipeOut .
- [Asterisk-Users] Weird experience with MOH
WipeOut .
- [Asterisk-Users] New budgetone firmware
WipeOut .
- [Asterisk-Users] New budgetone firmware
WipeOut .
- [Asterisk-Users] Budgetone Transfer (The answer)
WipeOut .
- [Asterisk-Users] addmailbox2 (Attached)
WipeOut .
- [Asterisk-Users] grandstream sip phone
WipeOut .
- [Asterisk-Users] grandstream sip phone
WipeOut .
- [Asterisk-Users] AVM Fritz! to connect LAN with ISDN line?
WipeOut .
- [Asterisk-Users] AVM Fritz! to connect LAN with ISDN line?
WipeOut .
- [Asterisk-Users] Grandstream BudgeTone 102 initial experiences
WipeOut .
- [Asterisk-Users] Budgetone and NTP (redux)
WipeOut .
- [Asterisk-Users] UK call termination..
WipeOut .
- [Asterisk-Users] ADPCM Codec
WipeOut .
- [Asterisk-Users] *--IAX--* problems.
WipeOut .
- [Asterisk-Users] *--IAX--* problems. (chan_capi problem)
WipeOut .
- [Asterisk-Users] 2 B channels for ISDN cards
WipeOut .
- [Asterisk-Users] Asterisk /SIP .. nat
WipeOut .
- [Asterisk-Users] can't get musiconhold to work
WipeOut .
- [Asterisk-Users] can't get musiconhold to work
WipeOut .
- [Asterisk-Users] can't get musiconhold to work
WipeOut .
- [Asterisk-Users] Asterisk SIP + Grandstream 100 phone
WipeOut .
- [Asterisk-Users] ISDN Fritz & RedHat 8.0
WipeOut .
- [Asterisk-Users] g729 Codec
WipeOut .
- [Asterisk-Users] ISDN Fritz & RedHat 8.0
WipeOut .
- [Asterisk-Users] Asterisk user guide ..
WipeOut .
- [Asterisk-Users] Crossover T1 cable
Iván Aponte
- [Asterisk-Users] FXS and PBX Integration
Iván Aponte
- [Asterisk-Users] OT: list format vs newsgroup format
Iván Aponte
- [Asterisk-Users] RTP session traversing Asterisk server ...
Juan Heriberto Brito Jiménez
- [Asterisk-Users] Chan_capi hanging channels
Petr Michálek
- [Asterisk-Users] How to modify dialed number?
Petr Michálek
- [Asterisk-Users] picking up a ringing extension
carlos del mayor
- [Asterisk-Users] modules.conf
carlos del mayor
- [Asterisk-Users] callgroup and pickupgroup
carlos del mayor
- [Asterisk-Users] register line on sip.conf
carlos del mayor
- [Asterisk-Users] Transfert call
carlos del mayor
- [Asterisk-Users] Transfert call
carlos del mayor
- [Asterisk-Users] chanh323 dialling
carlos del mayor
- [Asterisk-Users] oh323 problem (small one)
carlos del mayor
- [Asterisk-Users] Transfert call
carlos del mayor
- [Asterisk-Users] Using multiple iconnecthere accounts
carlos del mayor
- [Asterisk-Users] modules.conf again
carlos del mayor
- [Asterisk-Users] error on web page for msn
carlos del mayor
- [Asterisk-Users] modules.conf again
carlos del mayor
- [Asterisk-Users] Asterix Manual
carlos del mayor
- [Asterisk-Users] more abou msn
carlos del mayor
- [Asterisk-Users] more abou msn
carlos del mayor
- [Asterisk-Users] What does "callerid=" in sip.conf do?
carlos del mayor
- [Asterisk-Users] Problem with echo
Daniel ANDRE
- [Asterisk-Users] Linejack strikes again.
Andres Tello Abrego
- [Asterisk-Users] Newbie - Looking for pointers
Adams, Gavin
- [Asterisk-Users] Zaptel cards, working FXS and SIP, no audio?
Adams, Gavin
- [Asterisk-Users] Client Call Management Application?
Marcus Adolfsson
- [Asterisk-Users] Stream Phone Call: Sound on Consule OSS to Helix server?
Marcus Adolfsson
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
Marcus Adolfsson
- [Asterisk-Users] Sip call pickup ?
Tan Aks
- [Asterisk-Users] Problem with echo
Tan Aks
- [Asterisk-Users] Asterisk and Hot Desks??
Tan Aks
- [Asterisk-Users] Accurate Billing
Tan Aks
- [Asterisk-Users] Direct entry to your own voice mailbox
Tan Aks
- [Asterisk-Users] caller id
Tan Aks
- [Asterisk-Users] New budgetone firmware
Tan Aks
- [Asterisk-Users] SIP Authentication bug?
Tan Aks
- [Asterisk-Users] UK call termination..
Tan Aks
- [Asterisk-Users] Grandstream Budgettone 100 & 102
Tan Aks
- [Asterisk-Users] module : cdr_sybase.so
Chris Albertson
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality speech synthesis?
Chris Albertson
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality speech synthesis?
Chris Albertson
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality speech synthesis?
Chris Albertson
- [Asterisk-Users] Text to Speech - Someone needs to do this
Chris Albertson
- [Asterisk-Users] Text to Speech - Someone needs to do this
Chris Albertson
- [Asterisk-Users] A solution for SIP and NAT
Richard Alexander
- [Asterisk-Users] Cisco 7960s
Erik Anderson
- [Asterisk-Users] New Member
Erik Anderson
- [Asterisk-Users] Asterisk and VMWare
Erik Anderson
- [Asterisk-Users] module : cdr_sybase.so
Erik Anderson
- [Asterisk-Users] cdr_mysql
Erik Anderson
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
Erik Anderson
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Erik Anderson
- [Asterisk-Users] iaxclient (Activex)
Erik Anderson
- [Asterisk-Users] Configuration
Erik Anderson
- [Asterisk-Users] Compilations errors
Erik Anderson
- [Asterisk-Users] Microsoft SQL: cdr_tds.c
Erik Anderson
- [Asterisk-Users] codec problems with asterisk
Reece Anderson
- [Asterisk-Users] SetLanguage application doesn;t seem to work in latest Asterisk
Panagidou Anna
- [Asterisk-Users] BIG problem with multiple rings before pickup
Joe Antkowiak
- [Asterisk-Users] Newbie Doubts
Joe Antkowiak
- [Asterisk-Users] PCI Master Abort
Joe Antkowiak
- [Asterisk-Users] serious dtmf recognition problem.
Joe Antkowiak
- [Asterisk-Users] serious dtmf recognition problem.
Joe Antkowiak
- [Asterisk-Users] serious dtmf recognition problem.
Joe Antkowiak
- [Asterisk-Users] H3500CW recommendation
Joe Antkowiak
- [Asterisk-Users] Asterisk and FWD
Joe Antkowiak
- [Asterisk-Users] Ideal Prompt Recording Setup?
Joe Antkowiak
- [Asterisk-Users] Supplementing Current phone system
Joe Antkowiak
- [Asterisk-Users] IRQ Misses?
Joe Antkowiak
- [Asterisk-Users] SCO/Linux concerns
Joe Antkowiak
- [Asterisk-Users] BIG problem with multiple rings before pickup
Jim Archer
- [Asterisk-Users] BIG problem with multiple rings before pickup
Jim Archer
- [Asterisk-Users] BIG problem with multiple rings before pickup
Jim Archer
- [Asterisk-Users] BIG problem with multiple rings before pickup
Jim Archer
- [Asterisk-Users] BIG problem with multiple rings before pickup
Jim Archer
- [Asterisk-Users] BIG problem with multiple rings before pickup
Jim Archer
- [Asterisk-Users] BIG problem with multiple rings before pickup
Jim Archer
- [Asterisk-Users] No ringing when I dial an extension
Jim Archer
- [Asterisk-Users] Segmentation fault with chan_oh323
Arun Kumar Sharma, Noida
- [Asterisk-Users] [Asterisk-Users]Help Needed
Arun Kumar Sharma, Noida
- [Asterisk-Users] Help Needed
Arun Kumar Sharma, Noida
- [Asterisk-Users] Help Needed
Arun Kumar Sharma, Noida
- [Asterisk-Users] Asterisk against 3 Com NBX 100 and
Siemens HiPath 3700/3750
Asterisk
- [Asterisk-Users] Network design question
Asterisk
- [Asterisk-Users] ATA-186 de-register
Humberto Atristain
- [Asterisk-Users] PAnasonic And Asterisk
Humberto Atristain
- [Asterisk-Users] MGCP behind NAT
Humberto Atristain
- [Asterisk-Users] Vonage
Humberto Atristain
- [Asterisk-Users] Loop Drop on vpb/1-7
Martin Atukunda
- [Asterisk-Users] Conferences with CAPI and H323
Rattana BIV
- [Asterisk-Users] Transfert call
Rattana BIV
- [Asterisk-Users] Transfert call
Rattana BIV
- [Asterisk-Users] Matching winth asterisk-oh323
Rattana BIV
- [Asterisk-Users] Compilations errors
Rattana BIV
- [Asterisk-Users] Some stats
Rattana BIV
- [Asterisk-Users] Manager
Rattana BIV
- [Asterisk-Users] Sound Quality.
Michael Baird
- [Asterisk-Users] Rhino Channel Bank,
Yoanes Bandung
- [Asterisk-Users] Remote Agents
Derek Barber
- [Asterisk-Users] Remote Agents
Derek Barber
- [Asterisk-Users] BudgeTone-100 Early Dial
Paul Barrett
- [Asterisk-Users] BudgeTone-100 Early Dial
Paul Barrett
- [Asterisk-Users] BudgeTone-100 Date and Time
Paul Barrett
- [Asterisk-Users] New budgetone firmware
Paul Barrett
- [Asterisk-Users] Welltech FXS SIP registering with Asterisk
Elliott Bay
- [Asterisk-Users] Can't access outside voicemail services through asterisk
Derek Beaumont
- [Asterisk-Users] PCI Master Abort
Derek Beaumont
- [Asterisk-Users] Using multiple iconnecthere accounts
Derek Beaumont
- [Asterisk-Users] Using multiple iconnecthere accounts
Derek Beaumont
- [Asterisk-Users] Using multiple iconnecthere accounts
Derek Beaumont
- [Asterisk-Users] Using multiple iconnecthere accounts
Derek Beaumont
- [Asterisk-Users] sip jitter buffer
Derek Beaumont
- [Asterisk-Users] TDM10B - Dies after a few hours
Derek Beaumont
- [Asterisk-Users] Unable to find IP address???
Derek Beaumont
- [Asterisk-Users] Conditional Contexts
Derek Beaumont
- [Asterisk-Users] Conditional Contexts
Derek Beaumont
- [Asterisk-Users] Conditional Contexts
Derek Beaumont
- [Asterisk-Users] outgoing callerid string
Derek Beaumont
- [Asterisk-Users] TDM10B - Dies after a few hours
Brad Bergman
- [Asterisk-Users] TDM10B - Dies after a few hours
Brad Bergman
- [Asterisk-Users] DTMF control for TDM device?
Brad Bergman
- [Asterisk-Users] the 'pound' and '#' are the same?
Brad Bergman
- [Asterisk-Users] voicemail enhancements
Brad Bergman
- [Asterisk-Users] VoiceMail2 Wish List
Brad Bergman
- [Asterisk-Users] VoiceMail2 Wish List
Brad Bergman
- [Asterisk-Users] BudgeTone-100 Early Dial
Stephen R. Besch
- [Asterisk-Users] grandstream sip phone (NTP)
Stephen R. Besch
- [Asterisk-Users] Budgetone and NTP (redux)
Stephen R. Besch
- [Asterisk-Users] Best software SIP client
Stephen R. Besch
- [Asterisk-Users] Logfile rotation
Michael Bielicki
- [Asterisk-Users] Asterisk Sacrifice?
Michael Bielicki
- [Asterisk-Users] asterisk-oh323 v0.5.3
Michael Bielicki
- [Asterisk-Users] Budgetone and Voicemail
Michael Bielicki
- [Asterisk-Users] TDM10B - Dies after a few hours
Michael Bielicki
- [Asterisk-Users] asterisk and modem
Michael Bielicki
- [Asterisk-Users] Remote Agents
Michael Bielicki
- [Asterisk-Users] payload framesize
Michael Bielicki
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality speech synthesis?
Michael Bielicki
- [Asterisk-Users] g729 + oh323
Michael Bielicki
- [Asterisk-Users] TE410P startup
Michael Bielicki
- [Asterisk-Users] Zaptel
Michael Bielicki
- [Asterisk-Users] Call Transfer, Budgettone 100
Michael Bielicki
- [Asterisk-Users] Setting up A TDM400P
John Bigelow
- [Asterisk-Users] Asteriks, GnuGk and outgoing calls
Mickey Binder
- [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything
Mickey Binder
- [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything
Mickey Binder
- [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything
Mickey Binder
- [Asterisk-Users] Asterisk takes over
Sam Bingner
- [Asterisk-Users] chan_h323.c compile error
Bisker, Scott (7805)
- [Asterisk-Users] H.323 CallerID
Bisker, Scott (7805)
- [Asterisk-Users] Web Based Frontend
Chris Bond
- [Asterisk-Users] PCI Master Abort
Leo Ann Boon
- [Asterisk-Users] Budgetone and Voicemail
Brian Borders
- [Asterisk-Users] Budgetone and Voicemail
Brian Borders
- [Asterisk-Users] Can I interoperate with public PSTN gateways ?
David Boreham
- [Asterisk-Users] Can I interoperate with public PSTN gateways ?
David Boreham
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
David Boreham
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
David Boreham
- [Asterisk-Users] file repository
Steve Bourg
- [Asterisk-Users] picking up a ringing extension
Matteo Brancaleoni
- [Asterisk-Users] A solution for SIP and NAT
Matteo Brancaleoni
- [Asterisk-Users] record a conversation
Matteo Brancaleoni
- [Asterisk-Users] res parking patch
Matteo Brancaleoni
- [Asterisk-Users] res parking patch
Matteo Brancaleoni
- [Asterisk-Users] Activate MySQL logging
Matteo Brancaleoni
- [Asterisk-Users] Activate MySQL logging
Matteo Brancaleoni
- [Asterisk-Users] Accurate Billing
Matteo Brancaleoni
- [Asterisk-Users] New budgetone firmware
Matteo Brancaleoni
- [Asterisk-Users] SIP info
Matteo Brancaleoni
- [Asterisk-Users] h323 and oh323 modules
Matteo Brancaleoni
- [PATCH] [Asterisk-Users] SIP info
Matteo Brancaleoni
- [Asterisk-Users] Need help with DTMF issues and IVR systems
Matteo Brancaleoni
- [Asterisk-Users] SIP Call Forwarding/Transfer support ?
Hielke Christian Braun
- AW: [Asterisk-Users] * behind ISDN pbx - Forwarding to extensions with in primary pbx
Olga und Andreas Brodowski
- AW: AW: [Asterisk-Users] * behind ISDN pbx - Forwarding to extensions with in primary pbx
Olga und Andreas Brodowski
- AW: [Asterisk-Users] Microsoft SQL
Olga und Andreas Brodowski
- [Asterisk-Users] .gsm voice format
Adrian Brown
- [Asterisk-Users] Best E1 channel bank?
Adrian Brown
- [Asterisk-Users] [Q]: Dialin problems over E1 on a Digium E100P
Chris Bshaw
- [Asterisk-Users] Unsubscribe
Krzysztof Bujak
- [Asterisk-Users] BIG problem with multiple rings before pickup
Ryan Butler
- [Asterisk-Users] Transfers on the Cisco 7960
Ryan Butler
- [Asterisk-Users] OT: list format vs newsgroup format
Ryan Butler
- [Asterisk-Users] Asterisk NOOB
CTI
- [Asterisk-Users] questions
CTI
- [Asterisk-Users] Asterisk PBX Billing
Kim C. Callis
- [Asterisk-Users] ATA-186 de-register
Kim C. Callis
- [Asterisk-Users] Asterisk PBX Billing
Kim C. Callis
- [Asterisk-Users] LD accontability
Kim C. Callis
- [Asterisk-Users] Asterisk Sacrifice?
Kim C. Callis
- [Asterisk-Users] Accurate Billing
Kim C. Callis
- [Asterisk-Users] Using Efax for virtual fax?
Kim C. Callis
- [Asterisk-Users] Transfers on the Cisco 7960
Kim C. Callis
- [Asterisk-Users] Any Carrier Accessbank I users
Kim C. Callis
- [Asterisk-Users] Analog features over the ATA-186
Kim C. Callis
- [Asterisk-Users] Any dialing tricks...
Kim C. Callis
- [Asterisk-Users] cdr_mysql
Kim C. Callis
- [Asterisk-Users] QoS for Asterisk
Kim C. Callis
- [Asterisk-Users] Codecs for use with Cisco 7960 and ATA-186
Kim C. Callis
- [Asterisk-Users] Codecs for use with Cisco 7960 and ATA-186
Kim C. Callis
- [Asterisk-Users] Codecs for use with Cisco 7960 and ATA-186
Kim C. Callis
- [Asterisk-Users] Database usage?
Kim C. Callis
- [Asterisk-Users] meetme room
Kim C. Callis
- [Asterisk-Users] A solution for SIP and NAT
Michael C. Cambria
- [Asterisk-Users] DTMF control for TDM device?
Brian Capouch
- [Asterisk-Users] IAX and Speex?
Brian Capouch
- [Asterisk-Users] Music on hold & Read error on sound device
Brian Capouch
- [Asterisk-Users] QoS for Asterisk
Brian Capouch
- [Asterisk-Users] Cisco 802.11b VoIP phone?
Brian Capouch
- [Asterisk-Users] Changes to reset method for ATA186?
Brian Capouch
- [Asterisk-Users] time and date stamp in voicemail
Brian Capouch
- [Asterisk-Users] Nortel 350
Brian Capouch
- [Asterisk-Users] Cisco 7960
William Carlson
- [Asterisk-Users] Cisco 7960
William Carlson
- [Asterisk-Users] Cisco 7960
William Carlson
- [Asterisk-Users] Voicemail2 unified messaging
David Carr
- [Asterisk-Users] Wildcard E100P resellers in Europe ?
Nicolas Cartron
- [Asterisk-Users] Wildcard E100P resellers in Europe ?
Nicolas Cartron
- [Asterisk-Users] chanh323 dialling
Dave Alan Caruana
- [Asterisk-Users] re. rtp.c RTP codec 19
Dave Alan Caruana
- [Asterisk-Users] RTP.C codec error 19
Dave Alan Caruana
- [Asterisk-Users] oh323 problem (small one)
Dave Alan Caruana
- [Asterisk-Users] oh323 prob :)
Dave Alan Caruana
- [Asterisk-Users] SIP disconnecting : response 481
Dave Alan Caruana
- [Asterisk-Users] SIP Problem (previous post) .. information might be relevant
Dave Alan Caruana
- [Asterisk-Users] -- Got SIP response 481 "Invalid CSeq Number" back from 216.52.153.207
Dave Alan Caruana
- [Asterisk-Users] OH323 + G729 + Go2Call
Dave Alan Caruana
- [Asterisk-Users] OH323 + G729 + Go2Call
Dave Alan Caruana
- [Asterisk-Users] ISDN PRI E1 configuration with E100P
Dave Alan Caruana
- [Asterisk-Users] G729 codec problems
Dave Alan Caruana
- [Asterisk-Users] Chan_H323, G729 (minor problem)
Dave Alan Caruana
- [Asterisk-Users] H323/No one is available to answer at this time
Dave Alan Caruana
- [Asterisk-Users] Chan_H323, G729 (minor problem)
Dave Alan Caruana
- [Asterisk-Users] interfacing asterisk with a legacy PBX
Dave Alan Caruana
- [Asterisk-Users] Asterisk user guide ..
Dave Alan Caruana
- [Asterisk-Users] stupid questions ..
Dave Alan Caruana
- [Asterisk-Users] stupid questions ..
Dave Alan Caruana
- [Asterisk-Users] stupid questions ..
Dave Alan Caruana
- [Asterisk-Users] SIP calls cause Segmentation Fault
Dave Alan Caruana
- [Asterisk-Users] Asterisk <--> TTS server
Cerrajetto
- [Asterisk-Users] RTP codec 13 received - Cisco incompatibility?
Cerrajetto
- [Asterisk-Users] Please help -- Syntax for dialing VoIP provider
Paul Cheng
- [Asterisk-Users] Please help -- Syntax for dialing VoIP provider
Paul Cheng
- [Asterisk-Users] IAXTEL toll-free
Paul Cheng
- [Asterisk-Users] Re: Strange behavior in latest CVS
Paul Cheng
- [Asterisk-Users] Need help with DTMF issues and IVR systems
Paul Cheng
- [Asterisk-Users] Need help with DTMF issues and IVR systems
Paul Cheng
- [Asterisk-Users] Asterisk - Protocol Converter from SIP/H.323
Lubomir Christov
- [Asterisk-Users] Using multiple iconnecthere accounts
Lubomir Christov
- [Asterisk-Users] Asterix Manual
Lubomir Christov
- [Asterisk-Users] H450 problems
Lubomir Christov
- [Asterisk-Users] what is wrong with gsm files
Lubomir Christov
- [Asterisk-Users] something is wrong with gsm prompts format
Lubomir Christov
- [Asterisk-Users] Phoneserve SIP provider
Lubomir Christov
- [Asterisk-Users] Phoneserve SIP provider
Lubomir Christov
- [Asterisk-Users] Help Needed
Lubomir Christov
- [Asterisk-Users] h323 and oh323 modules
Lubomir Christov
- [Asterisk-Users] msn
Kelvin Chua
- [Asterisk-Users] system alias
Kelvin Chua
- [Asterisk-Users] msn authentication
Kelvin Chua
- [Asterisk-Users] msn authentication
Kelvin Chua
- [Asterisk-Users] Question #3
Kelvin Chua
- [Asterisk-Users] grandstream sip phone
Kelvin Chua
- [Asterisk-Users] Cisco's CallManager and * (was: Cisco 7960g) (fwd)
Kelvin Chua
- [Asterisk-Users] audiocodes fxs
Kelvin Chua
- [Asterisk-Users] one way audio h323 callmanager
Kelvin Chua
- [Asterisk-Users] 24port or higher fxs
Kelvin Chua
- [Asterisk-Users] h323 Ringing sound
Jorge Cisneros
- [Asterisk-Users] Asterisk as SIP <-> PSTN gateway
Archie Cobbs
- [Asterisk-Users] Asterisk -> AS5300 SIP Interoperability
Daniel Concepcion
- [Asterisk-Users] Friendly Slow Faxing Reminder
John Congdon
- [Asterisk-Users] Enhanced queue app
John Congdon
- [Asterisk-Users] Agent in new CVS
John Congdon
- [Asterisk-Users] random hangups
John Congdon
- [Asterisk-Users] ADSI and SoftKeys
John Congdon
- [Asterisk-Users] Queue and Agents in CVS
John Congdon
- [Asterisk-Users] Music on hold quality..
Joe Cooke
- [Asterisk-Users] Grandstream Budgettone 100 & 102
Joe Cooke
- [Asterisk-Users] Vonage
Joe Cooke
- [Asterisk-Users] Minimum budget question ...
Stefano Corsi
- [Asterisk-Users] grandstream sip phone
Dave Cotton
- [Asterisk-Users] grandstream sip phone
Dave Cotton
- [Asterisk-Users] Gnophone
Dave Cotton
- [Asterisk-Users] the 'pound' and '#' are the same?
Dave Cotton
- [Asterisk-Users] the 'pound' and '#' are the same?
Dave Cotton
- [Asterisk-Users] SCO/Linux concerns
Dave Cotton
- [Asterisk-Users] Migration to Asterisk - Running off of Merlin Legend system
Steve Creel
- [Asterisk-Users] Vendors for phones
Steve Creel
- [Asterisk-Users] grandstream sip phone
Steve Creel
- [Asterisk-Users] FXS and PBX Integration
Steve Creel
- [Asterisk-Users] 7960 / MGCP
Steve Creel
- [Asterisk-Users] shared line-appearance
Steve Creel
- [Asterisk-Users] unsuscribe
Carlos Crembil
- [Asterisk-Users] %unsuscribe
Carlos Crembil
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Carlos Eduardo Cremon
- [Asterisk-Users] Answering on an zap device
Cristi
- [Asterisk-Users] Answering on an zap device
Cristi
- [Asterisk-Users] Answering on an zap device
Cristi
- [Asterisk-Users] Debug PRI!
Cristi
- [Asterisk-Users] Error on PRI channel : Call specified but not found!
Cristi
- [Asterisk-Users] Error on PRI channel : Call specified but not found!
Cristi
- [Asterisk-Users] ISDN PRI E1 configuration with E100P
Cristi
- [Asterisk-Users] Drop the call in 10min
Cristi
- [Asterisk-Users] Drop the call in 10min
Cristi
- [Asterisk-Users] Alphanumerical digits
Cristi
- [Asterisk-Users] Alphanumerical digits
Cristi
- [Asterisk-Users] error "WARNING[28697]: File app_dial.c, Line 304 (wait_for_answer):
Unable to forward voice"
Cristi
- [Asterisk-Users] Channel Language
Cristi
- [Asterisk-Users] Microsoft SQL
Cristi
- [Asterisk-Users] h323 over NAT
Cristi
- [Asterisk-Users] How do i make best use of Macro?
Steven Critchfield
- [Asterisk-Users] How do i make best use of Macro?
Steven Critchfield
- [Asterisk-Users] Conference calls
Steven Critchfield
- [Asterisk-Users] Conference calls
Steven Critchfield
- [Asterisk-Users] More switch => stuff
Steven Critchfield
- [Asterisk-Users] More switch => stuff
Steven Critchfield
- [Asterisk-Users] BIG problem with multiple rings before pickup
Steven Critchfield
- [Asterisk-Users] BIG problem with multiple rings before pickup
Steven Critchfield
- [Asterisk-Users] BIG problem with multiple rings before pickup
Steven Critchfield
- [Asterisk-Users] Is there any real asterisk documentation ?
Steven Critchfield
- [Asterisk-Users] Need a recommendation on a good
motherboard/processor combination
Steven Critchfield
- [Asterisk-Users] Drops due to codecs?
Steven Critchfield
- [Asterisk-Users] Migration to Asterisk - Running off of Merlin
Legend system
Steven Critchfield
- [Asterisk-Users] res parking patch
Steven Critchfield
- [Asterisk-Users] Coding Time/Date announce on voicemail
Steven Critchfield
- [Asterisk-Users] dst number
Steven Critchfield
- [Asterisk-Users] IVR problem from PSTN phone
Steven Critchfield
- [Asterisk-Users] Asterisk Sacrifice?
Steven Critchfield
- [Asterisk-Users] zt_pri_errors: PRI got event: 8 / 6
Steven Critchfield
- [Asterisk-Users] Coding Time/Date announce on voicemail
Steven Critchfield
- [Asterisk-Users] Asterisk Sacrifice?
Steven Critchfield
- [Asterisk-Users] Cllecting digits.
Steven Critchfield
- [Asterisk-Users] Integratting * With Database(Newbie)
Steven Critchfield
- [Asterisk-Users] E&M DID config question
Steven Critchfield
- [Asterisk-Users] TDM400P noise?
Steven Critchfield
- [Asterisk-Users] Digital phones
Steven Critchfield
- [Asterisk-Users] Digital phones
Steven Critchfield
- [Asterisk-Users] Accurate Billing
Steven Critchfield
- [Asterisk-Users] Accurate Billing
Steven Critchfield
- [Asterisk-Users] Accurate Billing
Steven Critchfield
- [Asterisk-Users] Asterisk and VMWare
Steven Critchfield
- [Asterisk-Users] Network design question
Steven Critchfield
- [Asterisk-Users] System command..
Steven Critchfield
- [Asterisk-Users] System command..
Steven Critchfield
- [Asterisk-Users] Can't access outside voicemail services
through asterisk
Steven Critchfield
- [Asterisk-Users] Need a recommendation on a good
motherboard/processor combination
Steven Critchfield
- [Asterisk-Users] Asterisk and VMWare
Steven Critchfield
- [Asterisk-Users] Dial plan doesn't seem to save properly
Steven Critchfield
- [Asterisk-Users] Lot's of errors and warnings.
Steven Critchfield
- [Asterisk-Users] Integratting * With Database(Newbie)
Steven Critchfield
- [Asterisk-Users] Asterisk and VMWare
Steven Critchfield
- [Asterisk-Users] Answering on an zap device
Steven Critchfield
- [Asterisk-Users] Answering on an zap device
Steven Critchfield
- [Asterisk-Users] Lot's of errors and warnings.
Steven Critchfield
- [Asterisk-Users] Answering on an zap device
Steven Critchfield
- [Asterisk-Users] Using multiple iconnecthere accounts
Steven Critchfield
- [Asterisk-Users] Lot's of errors and warnings.
Steven Critchfield
- [Asterisk-Users] line battery check
Steven Critchfield
- [Asterisk-Users] Debug PRI!
Steven Critchfield
- [Asterisk-Users] Call Accounting
Steven Critchfield
- [Asterisk-Users] Call Accounting
Steven Critchfield
- [Asterisk-Users] TDM400P noise?
Steven Critchfield
- [Asterisk-Users] Music on hold quality..
Steven Critchfield
- [Asterisk-Users] Music on hold quality..
Steven Critchfield
- [Asterisk-Users] Music on hold quality..
Steven Critchfield
- [Asterisk-Users] E1-RJ45 pin configuration
Steven Critchfield
- [Asterisk-Users] IAX G729 Codec
Steven Critchfield
- [Asterisk-Users] IAX G729 Codec
Steven Critchfield
- [Asterisk-Users] IAX G729 Codec
Steven Critchfield
- [Asterisk-Users] IAX G729 Codec
Steven Critchfield
- [Asterisk-Users] Using Efax for virtual fax?
Steven Critchfield
- [Asterisk-Users] T1 config for robbed-bit E&M AMI
Steven Critchfield
- [Asterisk-Users] Channel Bank configuration
Steven Critchfield
- [Asterisk-Users] Channel Bank configuration
Steven Critchfield
- [Asterisk-Users] IAX G729 Codec
Steven Critchfield
- [Asterisk-Users] channel bank configuration
Steven Critchfield
- [Asterisk-Users] channel bank configuration
Steven Critchfield
- [Asterisk-Users] channel bank configuration
Steven Critchfield
- [Asterisk-Users] New Member
Steven Critchfield
- [Asterisk-Users] Asterisk and VMWare
Steven Critchfield
- [Asterisk-Users] Line Override Device
Steven Critchfield
- [Asterisk-Users] unresolved symbols in
/lib/modules/2.4.18/misc/zaptel.o
Steven Critchfield
- [Asterisk-Users] Line Override Device
Steven Critchfield
- [Asterisk-Users] Line Override Device
Steven Critchfield
- [Asterisk-Users] .gsm voice format
Steven Critchfield
- AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30%
System)
Steven Critchfield
- [Asterisk-Users] Asterisk and VMWare
Steven Critchfield
- [Asterisk-Users] .gsm voice format
Steven Critchfield
- [Asterisk-Users] EZ-Install
Steven Critchfield
- X100P mod or USB relay box, RE: [Asterisk-Users] Line Override
Device
Steven Critchfield
- [Asterisk-Users] .gsm voice format
Steven Critchfield
- [Asterisk-Users] EZ-Install
Steven Critchfield
- [Asterisk-Users] Getting started
Steven Critchfield
- [Asterisk-Users] audio pause/delay problems
Steven Critchfield
- X100P mod or USB relay box, RE: [Asterisk-Users] Line Override
Device
Steven Critchfield
- [Asterisk-Users] Hardware Vendors
Steven Critchfield
- [Asterisk-Users] ebs mbs p-phone question
Steven Critchfield
- [Asterisk-Users] insmod wcfxo failed ( b8zs, esf, wink start
is what I'm trying to do.)
Steven Critchfield
- [Asterisk-Users] Conditional Contexts
Steven Critchfield
- [Asterisk-Users] Stream Phone Call: Sound on Consule OSS to
Helix server?
Steven Critchfield
- [Asterisk-Users] Receptionist Station
Steven Critchfield
- [Asterisk-Users] Asterisk on Cygwin?
Steven Critchfield
- [Asterisk-Users] Vendors for phones
Steven Critchfield
- [Asterisk-Users] Asterisk on Cygwin?
Steven Critchfield
- [Asterisk-Users] Multiple Phones for 1 Extension
Steven Critchfield
- [Asterisk-Users] Any dialing tricks...
Steven Critchfield
- [Asterisk-Users] Silly questions due to ingrained knowledge of
analog phone use.
Steven Critchfield
- [Asterisk-Users] AGI & Silence detection
Steven Critchfield
- [Asterisk-Users] Call Parking
Steven Critchfield
- [Asterisk-Users] serious dtmf recognition problem.
Steven Critchfield
- [Asterisk-Users] H3500CW recommendation
Steven Critchfield
- [Asterisk-Users] OT: list format vs newsgroup format
Steven Critchfield
- [Asterisk-Users] serious dtmf recognition problem.
Steven Critchfield
- [Asterisk-Users] OT: list format vs newsgroup format
Steven Critchfield
- [Asterisk-Users] H3500CW recommendation
Steven Critchfield
- [Asterisk-Users] OT: list format vs newsgroup format
Steven Critchfield
- [Asterisk-Users] H3500CW recommendation
Steven Critchfield
- [Asterisk-Users] Asterisk NOOB
Steven Critchfield
- [Asterisk-Users] questions
Steven Critchfield
- [Asterisk-Users] questions
Steven Critchfield
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Steven Critchfield
- [Asterisk-Users] Self parked but avaliable
Steven Critchfield
- [Asterisk-Users] Music on hold & Read error on sound device
Steven Critchfield
- [off-list] Re: [Asterisk-Users] G729 doesn't work.
Steven Critchfield
- [Asterisk-Users] Integrating cell phone into Asterisk
Extension..
Steven Critchfield
- [Asterisk-Users] Outgoing calls through a calling card
Steven Critchfield
- [Asterisk-Users] newbie - simple dialout server
Steven Critchfield
- [Asterisk-Users] how do I do s extensions with PRI
Steven Critchfield
- [Asterisk-Users] Integrating cell phone into Asterisk
Extension..
Steven Critchfield
- [Asterisk-Users] compilation error
Steven Critchfield
- [Asterisk-Users] Asterisk <--> TTS server
Steven Critchfield
- [Asterisk-Users] time and date stamp in voicemail
Steven Critchfield
- [Asterisk-Users] Voicemail() problems - Long pause after
incoming message recording ended.
Steven Critchfield
- [Asterisk-Users] Ordering digital trunks?
Steven Critchfield
- [Asterisk-Users] Asterisk Developer's Kit (TDM) help
Steven Critchfield
- [Asterisk-Users] Microsoft SQL
Steven Critchfield
- [Asterisk-Users] Some stats
Steven Critchfield
- [Asterisk-Users] SCO/Linux concerns
Steven Critchfield
- [Asterisk-Users] Manager.pm port
Steven Critchfield
- [Asterisk-Users] Parking calls - why doesn't work?
Steven Critchfield
- [Asterisk-Users] Manager
Steven Critchfield
- [Asterisk-Users] Parking calls - why doesn't work?
Steven Critchfield
- [Asterisk-Users] Manager
Steven Critchfield
- [Asterisk-Users] H.323 Gateway Connection
Szymon Czyz
- [Asterisk-Users] Cisco 7960g
Yifang Dai
- [Asterisk-Users] Cisco 7960g
Yifang Dai
- [Asterisk-Users] Cisco's CallManager and * (was: Cisco 7960g)
Yifang Dai
- [Asterisk-Users] Cisco 7960g
Yifang Dai
- [Asterisk-Users] Cisco's CallManager and *
Yifang Dai
- [Asterisk-Users] Problem with echo
Dan
- [Asterisk-Users] Actiontec's InternetPhoneWizard and Asterisk
Dan
- [Asterisk-Users] Problem with echo
Dan
- [Asterisk-Users] Problem with echo
Dan
- [Asterisk-Users] Soft SIP phones (with RING !!)
Dan
- [Asterisk-Users] Newbie question
Dan
- [Asterisk-Users] Modem channel in Asterisk
Dan
- [Asterisk-Users] Newbie question
Dan
- [Asterisk-Users] PCI CARD
Dan
- [Asterisk-Users] Asterisk Sacrifice?
Dan
- [Asterisk-Users] Asterisk Sacrifice?
Dan
- [Asterisk-Users] Virtual fax on the Asterisk box
Dan
- [Asterisk-Users] Virtual fax on the Asterisk box
Dan
- [Asterisk-Users] Virtual fax on the Asterisk box
Dan
- [Asterisk-Users] Virtual fax on the Asterisk box
Dan
- [Asterisk-Users] Virtual fax on the Asterisk box
Dan
- [Asterisk-Users] Direct entry to your own voice mailbox
Dan
- [Asterisk-Users] Asterisk and VMWare
Dan
- [Asterisk-Users] Direct entry to your own voice mailbox
Dan
- [Asterisk-Users] Direct entry to your own voice mailbox
Dan
- [Asterisk-Users] Direct entry to your own voice mailbox
Dan
- [Asterisk-Users] Asterisk and VMWare
Dan
- [Asterisk-Users] Asterisk and VMWare
Dan
- [Asterisk-Users] Virtual fax on the Asterisk box
Dan
- [Asterisk-Users] Asterisk and VMWare
Dan
- [Asterisk-Users] Budgetone and Voicemail
Dan
- [Asterisk-Users] incoming callerid on FXO
Dan
- [Asterisk-Users] incoming callerid on FXO
Dan
- [Asterisk-Users] Cellphone as an exchange line
Dan
- [Asterisk-Users] Cisco 7960 SIP Craziness...
Dan
- [Asterisk-Users] Asterisk and VMWare
Dan
- [Asterisk-Users] Asterisk and VMWare
Dan
- [Asterisk-Users] Asterisk and VMWare
Dan
- [Asterisk-Users] Asterisk and VMWare
Dan
- [Asterisk-Users] .gsm voice format
Dan
- [Asterisk-Users] Asterisk and VMWare
Dan
- [Asterisk-Users] voicemail instructions
Dan
- [Asterisk-Users] voicemail instructions
Dan
- [Asterisk-Users] Cisco 7960
Dan
- [Asterisk-Users] Correct syntax to call using IAX and a different UDP port
Dan
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
Dan
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Dan
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Dan
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Dan
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Dan
- [Asterisk-Users] IAX can be used on a different UDP port?
Dan
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Dan
- [Asterisk-Users] Actiontec's InternetPhoneWizard (USB) and Asterisk
Dan
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Dan
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Dan
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Dan
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
Dan
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Dan
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Dan
- [Asterisk-Users] anyone with X100P & Callerid working outside US ?
Dan
- [Asterisk-Users] iConnect Here PCPhone application and Asterisk
Dan
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Dan
- [Asterisk-Users] Asterisk and FWD
Dan
- [Asterisk-Users] Asterisk and FWD
Dan
- [Asterisk-Users] Cisco 802.11b VoIP phone?
Dan
- [Asterisk-Users] X-Lite Build 1016
Dan
- [Asterisk-Users] Codecs for use with Cisco 7960 and ATA-186
Dan
- [Asterisk-Users] iaxclient (Activex)
Dan
- [Asterisk-Users] Cisco ATA Advanced CallerID
Dan
- [Asterisk-Users] IAX and Call format
Dan
- [Asterisk-Users] time and date stamp in voicemail
Dan
- [Asterisk-Users] time and date stamp in voicemail
Dan
- [Asterisk-Users] can't get musiconhold to work
Dan
- [Asterisk-Users] moh/playback for non-zap interfaces
Dan
- [Asterisk-Users] PCM Voice Quality Issue on CVS Version
Dan
- [Asterisk-Users] Call transfer between two phones on the same ATA
Dan
- [Asterisk-Users] Call transfer on ATA186
Dan
- [Asterisk-Users] Call transfer on ATA186
Dan
- [Asterisk-Users] Call transfer on ATA186
Dan
- [Asterisk-Users] Call transfer on ATA186
Dan
- [Asterisk-Users] RTP session traversing Asterisk server...
Dan
- [Asterisk-Users] Call transfer on ATA186
Dan
- [Asterisk-Users] Call Transfer
Dan
- [Asterisk-Users] Voicemail message forwarded to another extension and file format changing
Dan
- [Asterisk-Users] Call Transfer
Dan
- [Asterisk-Users] Call Transfer
Dan
- [Asterisk-Users] Call Transfer
Dan
- [Asterisk-Users] Dummy account/extension
Dan
- [Asterisk-Users] Dummy account/extension
Dan
- [Asterisk-Users] Dummy account/extension - Workaround for attended call trabsfer to ATA186
Dan
- [Asterisk-Users] Voicemail message forwarded to another extension and file format changing
Dan
- [Asterisk-Users] Dummy account/extension
Dan
- [Asterisk-Users] rxgain and txgain in zapata.conf
Dan
- [Asterisk-Users] X-Lite and Call transfer using Asterisk
Dan
- [Asterisk-Users] X-Lite and Call transfer using Asterisk
Dan
- [Asterisk-Users] Parking calls - why doesn't work?
Dan
- [Asterisk-Users] Manager
Dan
- [Asterisk-Users] Parking calls - why doesn't work?
Dan
- [Asterisk-Users] 'System' application exit with error even if it performs the job as expected
Dan
- [Asterisk-Users] 'System' application exit with error even if it performs the job as expected
Dan
- [Asterisk-Users] 'System' application exit with error even if it performs the job as expected
Dan
- [Asterisk-Users] Parking calls - why doesn't work?
Dan
- [Asterisk-Users] Parking calls - why doesn't work?
Dan
- [Asterisk-Users] 'System' application exit with error even if it performs the job as expected - partially solved
Dan
- [Asterisk-Users] 'System' application exit with error even if it performs the job as expected
Dan
- [Asterisk-Users] 'System' application exit with error even if it performs the job as expected - partially solved
Dan
- [Asterisk-Users] Seg Fault!!
Marian Danisek
- [Asterisk-Users] asterisk and uclinux
Marian Danisek
- [Asterisk-Users] dbget & dbput
Marian Danisek
- [Asterisk-Users] caller id
Marian Danisek
- [Asterisk-Users] caller id
Marian Danisek
- [Asterisk-Users] connect 2 asterisk boxes
Marian Danisek
- [Asterisk-Users] grandstream sip phone
Marian Danisek
- [Asterisk-Users] chan_capi error
Marian Danisek
- [Asterisk-Users] A solution for SIP and NAT
Klaus Darilion
- [Asterisk-Users] MSN Messenger 4.7 vs 5.0
Klaus Darilion
- [Asterisk-Users] Text to Speech - Someone needs to do this
Matthew John Darnell
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality speech synthesis?
Matthew John Darnell
- [Asterisk-Users] MSN Messenger(4.7) Setup
Neel Datta
- [Asterisk-Users] MSN Messenger(4.7) Setup
Neel Datta
- [Asterisk-Users] Receptionist Station
Dave
- [Asterisk-Users] Analog phone not ringing
Stephen Davies
- [Asterisk-Users] 'Echo' - I'm sure a common topic
Stephen Davies
- [Asterisk-Users] URGENT! brandly new Wildcard E400P for sale at $1000
Kalin Dikov
- [Asterisk-Users] three way calling and cisco ata 186
Thomas Dingermann
- [Asterisk-Users] three way calling and cisco ata 186
Thomas Dingermann
- [Asterisk-Users] mgcp problems
Thomas Dingermann
- [Asterisk-Users] Call transfer on ATA186
Thomas Dingermann
- [Asterisk-Users] Line Override Device
Shawn L. Djernes
- [Asterisk-Users] Line Override Device
Shawn L. Djernes
- [Asterisk-Users] Line Override Device
Shawn L. Djernes
- [Asterisk-Users] OT: list format vs newsgroup format
Steve Dommett
- [Asterisk-Users] SIP calls cause Segmentation Fault
Adam Donnison
- [Asterisk-Users] Zap outgoing hangups callprogress=yes?
Surfer Dude
- [Asterisk-Users] Asterisk as a stand alone voice mail server
Ronnie Earle
- [Asterisk-Users] H.323 Gateway Connection
Justin Eckhouse
- [Asterisk-Users] No Sound via Sip Phone
Justin Eckhouse
- [Asterisk-Users] Cisco 7960 Transfer Call drop problem
Justin Eckhouse
- [Asterisk-Users] Multiple Phones for 1 Extension
Justin Eckhouse
- [Asterisk-Users] Variable Substitution
Justin Eckhouse
- [Asterisk-Users] reconnecting
Darrell Eldridge
- [Asterisk-Users] Newbie Help
Thomas Elliott
- [Asterisk-Users] Newbie Help
Thomas Elliott
- [Asterisk-Users] X100P in Australia (was Asterisk-Users digest, Vol 1 #840 - 13 msgs)
Shaun Ewing
- [Asterisk-Users] Adtran TSU 600E
Jerk Face
- [Asterisk-Users] Call Dropping
Jerk Face
- [Asterisk-Users] Cisco ATA-186 config guide for Asterisk
Dan Fernandez
- [Asterisk-Users] Budgetone and Voicemail
Dan Fernandez
- [Asterisk-Users] Budgetone and Voicemail
Dan Fernandez
- [Asterisk-Users] Billsec on CDR
Dan Fernandez
- [Asterisk-Users] Delays with g729 and SIP. How come?
Dan Fernandez
- [Asterisk-Users] Problems with g729
Dan Fernandez
- [Asterisk-Users] Problems with g729
Dan Fernandez
- [Asterisk-Users] executing an agi script after a successful Dial
Dan Fernandez
- [Asterisk-Users] executing an agi script after a successful Dial
Dan Fernandez
- [Asterisk-Users] executing an agi script after a successful Dial
Dan Fernandez
- [Asterisk-Users] executing an agi script after a successful Dial
Dan Fernandez
- [Asterisk-Users] g729 Codec
Dan Fernandez
- [Asterisk-Users] RTP session traversing Asterisk server ...
Dan Fernandez
- [Asterisk-Users] iax2 and reinvites
Dan Fernandez
- [Asterisk-Users] Linejack strikes again.
Bruce Ferrell
- [Asterisk-Users] Using 2 PhoneJacks with Asterisk for Data calls.
Bruce Ferrell
- [Asterisk-Users] Problem with the Internet LineJACK ISA card...
Bruce Ferrell
- [Asterisk-Users] zt_pri_errors: PRI got event: 8 / 6
Stefano Finetti
- [Asterisk-Users] zt_pri_errors: PRI got event: 8 / 6
Stefano Finetti
- [Asterisk-Users] Remote * Using IAX
Stefano Finetti
- [Asterisk-Users] Getting Started with Digium T100/E100 cards
Stefano Finetti
- [Asterisk-Users] grandstream sip phone
Stefano Finetti
- [Asterisk-Users] grandstream sip phone
Stefano Finetti
- [Asterisk-Users] fxs without fxo
Stefano Finetti
- [Asterisk-Users] the 'pound' and '#' are the same?
Stefano Finetti
- [Asterisk-Users] ISDN Random Hangup Problems
Stefano Finetti
- [Asterisk-Users] A solution for SIP and NAT
Jim Flagg
- [Asterisk-Users] It's true - Nikotel charge for not-completed calls
Jim Flagg
- [Asterisk-Users] Drops due to codecs?
Daniel Flickinger
- [Asterisk-Users] Drops due to codecs?
Daniel Flickinger
- [Asterisk-Users] Offering an Asterisk Documentation and FAQ Portal
Damian Flynn
- [Asterisk-Users] Offering an Asterisk Documentation and FAQ Portal
Damian Flynn
- [Asterisk-Users] Connect 2 party with asterisk
Chee Foong
- [Asterisk-Users] g729 + oh323
Chee Foong
- [Asterisk-Users] g729 + oh323
Chee Foong
- [Asterisk-Users] g729 + oh323
Chee Foong
- [Asterisk-Users] Connect 2 call, any idea?
Chee Foong
- [Asterisk-Users] help on chan_h323
Chee Foong
- [Asterisk-Users] help on chan_h323
Chee Foong
- [Asterisk-Users] File chan_h323.c, Line 875
Chee Foong
- [Asterisk-Users] Call Transfer
Chee Foong
- [Asterisk-Users] Call Transfer
Chee Foong
- [Asterisk-Users] Call Transfer
Chee Foong
- [Asterisk-Users] Call Transfer
Chee Foong
- [Asterisk-Users] Call Transfer
Chee Foong
- [Asterisk-Users] Call Transfer
Chee Foong
- [Asterisk-Users] Call Transfer
Chee Foong
- [Asterisk-Users] sample.call + chan_h323 gives seg fault
Chee Foong
- [Asterisk-Users] sample.call + chan_h323 gives seg fault
Chee Foong
- [Asterisk-Users] Call Transfer
Chee Foong
- [Asterisk-Users] Need a recommendation on a good motherboard/processor combination
Mathew Frank
- [Asterisk-Users] Voice Modem + Soundcard Driver
Mathew Frank
- [Asterisk-Users] Enhanced queue app
Jim Friedeck
- [Asterisk-Users] Enhanced queue app
Jim Friedeck
- [Asterisk-Users] Enhanced queue app
Jim Friedeck
- [Asterisk-Users] Enhanced queue app
Jim Friedeck
- [Asterisk-Users] Enhanced queue app
Jim Friedeck
- [Asterisk-Users] Enhanced queue app
Jim Friedeck
- [Asterisk-Users] Enhanced queue app
Jim Friedeck
- [Asterisk-Users] Enhanced queue app
Jim Friedeck
- [Asterisk-Users] Enhanced queue app
Jim Friedeck
- [Asterisk-Users] Enhanced queue app
Jim Friedeck
- [Asterisk-Users] Need a recommendation on a good motherboard/processor
combination
Jim Friedeck
- [Asterisk-Users] Remote Agents
Jim Friedeck
- [Asterisk-Users] Remote Agents
Jim Friedeck
- [Asterisk-Users] Enhanced queue app
Jim Friedeck
- [Asterisk-Users] How do you force Asterisk to use only specific codecs?
Chip G
- [Asterisk-Users] How does Asterisk handle connecting two IP end points?
Chip G
- [Asterisk-Users] unsubscribe
Javi Gallart
- [Asterisk-Users] Asterisk and VMWare
Gary Gapinski
- [Asterisk-Users] Asterisk and VMWare
Gary Gapinski
- [Asterisk-Users] Compile Problems with gcc 3.3
Gary Gapinski
- [Asterisk-Users] the 'pound' and '#' are the same? (OT Rambling)
Gary Gapinski
- [Asterisk-Users] Problem with echo
Ing. Angel Gomez Garcia
- [Asterisk-Users] Problem with echo
Ing. Angel Gomez Garcia
- [Asterisk-Users] Conference calls
Ing. Angel Gomez Garcia
- [Asterisk-Users] Dialout Lines ???
Ing. Angel Gomez Garcia
- [Asterisk-Users] Cllecting digits.
Ing. Angel Gomez Garcia
- [Asterisk-Users] Cllecting digits.
Ing. Angel Gomez Garcia
- [Asterisk-Users] Integratting * With Database(Newbie)
Ing. Angel Gomez Garcia
- [Asterisk-Users] Cllecting digits.
Ing. Angel Gomez Garcia
- [Asterisk-Users] Cllecting digits.
Ing. Angel Gomez Garcia
- [Asterisk-Users] ECHO on sip- call
Ing. Angel Gomez Garcia
- [Asterisk-Users] experience with multi-port SIP/FXS gateways?
Ing. Angel Gomez Garcia
- [Asterisk-Users] Setting up A TDM400P
Ing. Angel Gomez Garcia
- [Asterisk-Users] audiocodes fxs
Ing. Angel Gomez Garcia
- [Asterisk-Users] audiocodes fxs
Ing. Angel Gomez Garcia
- [Asterisk-Users] Asterisk and Hot Desks??
Gary
- [Asterisk-Users] Asterisk PBX Billing
Gary
- [Asterisk-Users] error on web page for msn
Gary
- [Asterisk-Users] error on web page for msn
Gary
- [Asterisk-Users] X100P in Australia
Gary
- [Asterisk-Users] AUSTEL Certified
Gary
- [Asterisk-Users] Text to Speech - Someone needs to do this
Gary
- [Asterisk-Users] iaxclient (Activex)
Gary
- [Asterisk-Users] Direct Indial with ISDN and Netjet-S
Gary
- [Asterisk-Users] Direct Indial with ISDN and Netjet-S
Gary
- [Asterisk-Users] Newbie Doubts
Ricardo Saar Gemignani
- [Asterisk-Users] Newbie Doubts
Ricardo Saar Gemignani
- [Asterisk-Users] How does Asterisk handle connecting two IP end points?
Andrew Gillham
- [Asterisk-Users] Enhanced queue app
James Golovich
- [Asterisk-Users] Asterisk PBX Billing
James Golovich
- [Asterisk-Users] Call Recording
James Golovich
- [Asterisk-Users] IAX G729 Codec
Eduardo Goncalves
- [Asterisk-Users] E1 R2 on Asterisk
Eduardo Goncalves
- [Asterisk-Users] Linux flavor?
Eduardo Goncalves
- [Asterisk-Users] RFC2833 problems with X-Lite
Lee Goodman
- [Asterisk-Users] Back-to-back connected boards load test
Konrad Gorski
- [Asterisk-Users] PGSQL app and pbx parsing :-(
Adam Goryachev
- [Asterisk-Users] Problems with TDM40P
Adam Goryachev
- [Asterisk-Users] ATA 186 in Australia
Adam Goryachev
- FW: [Asterisk-Users] ATA 186 in Australia
Adam Goryachev
- [Asterisk-Users] Switch issues with non-dedicated comms.. (My experience)
Adam Goryachev
- [Asterisk-Users] Manager/gastman
Adam Goryachev
- [Asterisk-Users] AUSTEL Certified
Adam Goryachev
- [Asterisk-Users] Channel Banks
Adam Goryachev
- [Asterisk-Users] Australian Options
Adam Goryachev
- [Asterisk-Users] Australian Options
Adam Goryachev
- [Asterisk-Users] Australian Options
Adam Goryachev
- [Asterisk-Users] * Video changes
Jim Gottlieb
- [Asterisk-Users] FGB not waiting for digits
Jim Gottlieb
- [Asterisk-Users] FGB not waiting for digits
Jim Gottlieb
- [Asterisk-Users] BIG problem with multiple rings before pickup
Jim Gottlieb
- [Asterisk-Users] Virtual fax on the Asterisk box
Matthias Granberry
- [Asterisk-Users] Need a recommendation on a good motherboard/processor combination
Clay Graner
- [Asterisk-Users] Need a recommendation on a good motherboard/processor combination
Clay Graner
- [Asterisk-Users] Acom 200 B/D/E
Bradley Greep
- [Asterisk-Users] Dialout Lines ???
Bradley Greep
- [Asterisk-Users] interfacing asterisk with a legacy PBX
Bradley Greep
- [Asterisk-Users] How to make * send RTCP reports
HT
- [Asterisk-Users] How to make * send RTCP reports
HT
- [Asterisk-Users] Use dialing plan from h.323 gatekeeper?
HT
- [Asterisk-Users] Use dialing plan from h.323 gatekeeper?
HT
- [Asterisk-Users] g723.1 voicemail/conference files segfault *
HT
- [Asterisk-Users] g723.1 voicemail/conference files segfault *
HT
- AW: [Asterisk-Users] PGSQL app and pbx parsing :-(
Thomas Haeger
- [Asterisk-Users] CDR->dst and immediate=yes in zapata.conf
Thomas Haeger
- AW: [Asterisk-Users] Wildcard E100P resellers in Europe ?
Thomas Haeger
- [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)
Thomas Haeger
- AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)
Thomas Haeger
- AW: AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)
Thomas Haeger
- [Asterisk-Users] go on in context after the destination channel hung up?
Thomas Haeger
- [Asterisk-Users] go on in current context after destination channels hung up ?
Thomas Haeger
- AW: [Asterisk-Users] go on in current context after destination channels hung up ?
Thomas Haeger
- AW: [Asterisk-Users] go on in current context after destination channels hung up ?
Thomas Haeger
- [Asterisk-Users] Asterisk NOOB
Kyle Hagan
- [Asterisk-Users] Asterisk NOOB
Kyle Hagan
- [Asterisk-Users] Asterisk NOOB
Kyle Hagan
- [Asterisk-Users] Asterisk NOOB
Kyle Hagan
- [Asterisk-Users] Configuration
Kyle Hagan
- [Asterisk-Users] Best software SIP client
Kyle Hagan
- [Asterisk-Users] Asterisk Developer's Kit (TDM) help
Kyle Hagan
- [Asterisk-Users] CVS Problem?
Kyle Hagan
- [Asterisk-Users] CVS Problem?
Kyle Hagan
- [Asterisk-Users] Is there any real asterisk documentation ?
John Haigh
- [Asterisk-Users] Asterisk Sacrifice?
John Haigh
- [Asterisk-Users] Asterisk Call Manager doc
John Haigh
- [Asterisk-Users] Integrate Asterisk with Meridian phone system
John Haigh
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
Abdul Hakeem
- [Asterisk-Users] URGENT! brandly new Wildcard E400P for sale at $1000
Abdul Hakeem
- [Asterisk-Users] Offering an Asterisk Documentation and FAQ Portal
Abdul Hakeem
- [Asterisk-Users] RE: Voip Gateway Config
Abdul Hakeem
- [Asterisk-Users] Zaptel
Tais M. Hansen
- [Asterisk-Users] Zaptel
Tais M. Hansen
- [Asterisk-Users] Codecs
Tais M. Hansen
- [Asterisk-Users] SetCIDName
Tais M. Hansen
- [Asterisk-Users] SetCIDName
Tais M. Hansen
- [Asterisk-Users] Congestion
Tais M. Hansen
- [Asterisk-Users] Congestion
Tais M. Hansen
- [Asterisk-Users] Cisco 7960 And Firmware Upgrades
Matthew Hardeman
- [Asterisk-Users] Cisco 7960 SIP Craziness...
Matthew Hardeman
- [Asterisk-Users] Cisco 7960s
Matthew Hardeman
- [Asterisk-Users] Cisco 7960s
Matthew Hardeman
- [Asterisk-Users] Weird experience with MOH
Matthew Hardeman
- [Asterisk-Users] EZ-Install
Matthew Hardeman
- [Asterisk-Users] Hardware Vendors
Matthew Hardeman
- [Asterisk-Users] G729 licensing
Matthew Hardeman
- [Asterisk-Users] Fwd:[Vocal] Question about Cisco IP hard phones
Matthew Hardeman
- [Asterisk-Users] G729 licensing
Matthew Hardeman
- [Asterisk-Users] G729 quality
Matthew Hardeman
- [Asterisk-Users] Cisco 7960
Matthew Hardeman
- [Asterisk-Users] Any dialing tricks...
Matthew Hardeman
- [Asterisk-Users] Cisco 7960
Matthew Hardeman
- [Asterisk-Users] OT: list format vs newsgroup format
Matthew Hardeman
- [Asterisk-Users] OT: list format vs newsgroup format
Matthew Hardeman
- [Asterisk-Users] OT: list format vs newsgroup format
Matthew Hardeman
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Matthew Hardeman
- [Asterisk-Users] Asterisk crashes when trying to load G.729 module.
Matthew Hardeman
- [Asterisk-Users] How do you force Asterisk to use onlyspecific
codecs?
Vlasis Hatzistavrou
- [Asterisk-Users] Getting started
Johannes Herlitz
- [Asterisk-Users] TDM400P noise?
Kevin Herzig
- [Asterisk-Users] TDM400P noise?
Kevin Herzig
- [Asterisk-Users] VXML?
Kevin Herzig
- [Asterisk-Users] That is not a valid conference number meesage
Andy Hester
- [Asterisk-Users] Coding Time/Date announce on voicemail
Andy Hester
- [Asterisk-Users] Channel Bank configuration
Andy Hester
- [Asterisk-Users] Channel Bank configuration
Andy Hester
- [Asterisk-Users] New Member
Andy Hester
- [Asterisk-Users] time and date stamp in voicemail
Andy Hester
- [Asterisk-Users] time and date stamp in voicemail
Andy Hester
- [Asterisk-Users] time and date stamp in voicemail
Andy Hester
- [Asterisk-Users] time and date stamp in voicemail
Andy Hester
- [Asterisk-Users] time and date stamp in voicemail
Andy Hester
- [Asterisk-Users] time and date stamp in voicemail
Andy Hester
- [Asterisk-Users] Offering an Asterisk Documentation and FAQ Portal
Andy Hester
- [Asterisk-Users] RE Pingtel Phones
Andy Hester
- [Asterisk-Users] Best Analog sets for use w/*
Andy Hester
- [Asterisk-Users] Best Analog sets for use w/*
Andy Hester
- [Asterisk-Users] AUSTEL Certified
David Hindmarsh
- [Asterisk-Users] AUSTEL Certified
David Hindmarsh
- [Asterisk-Users] AGI & Silence detection
Stuart Hirst
- [Asterisk-Users] AGI & Silence detection
Stuart Hirst
- [Asterisk-Users] Music on hold & Read error on sound device
Stuart Hirst
- [Asterisk-Users] Music on hold & Read error on sound device
Stuart Hirst
- [Asterisk-Users] Music on hold & Read error on sound device
Stuart Hirst
- [Asterisk-Users] Best software SIP client
Stuart Hirst
- [Asterisk-Users] Music on hold & Read error on sound device
Stuart Hirst
- [Asterisk-Users] Music on hold & Read error on sound device
Stuart Hirst
- [Asterisk-Users] X-Lite Build 1016
Stuart Hirst
- [Asterisk-Users] Best software SIP client
Stuart Hirst
- [Asterisk-Users] Best software SIP client
Stuart Hirst
- [Asterisk-Users] Best software SIP client
Stuart Hirst
- [Asterisk-Users] Best software SIP client
Stuart Hirst
- [Asterisk-Users] Best software SIP client
Stuart Hirst
- [Asterisk-Users] ISDN Fritz & RedHat 8.0
Stuart Hirst
- [Asterisk-Users] ISDN Fritz & RedHat 8.0
Stuart Hirst
- [Asterisk-Users] ISDN Fritz & RedHat 8.0
Stuart Hirst
- [Asterisk-Users] ISDN Fritz & RedHat 8.0
Stuart Hirst
- [Asterisk-Users] CAPI & CLID
Stuart Hirst
- [Asterisk-Users] CAPI & CLID
Stuart Hirst
- [Asterisk-Users] X-Lite and Call transfer using Asterisk
Stuart Hirst
- [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #944 - 3 msgs
Mike Holloway
- [Asterisk-Users] ATA 186 in Australia
Steven Honson
- [Asterisk-Users] Cisco 7905G vs ATA186
Steven Honson
- [Asterisk-Users] PBX / Asterisk integration
Josh Howlett
- [Asterisk-Users] Cisco 7960s
Josh Howlett
- [Asterisk-Users] Newbe Questions.
Ing Isianto Istiadi
- [Asterisk-Users] IAX / MeetMe problem
JKNUTSEN at UP.COM
- [Asterisk-Users] MeetMe hangup
JKNUTSEN at UP.COM
- [Asterisk-Users] user agent with auto-pickup support - is there any?
Rahul Arvind Jadhav
- [Asterisk-Users] Asterisk /SIP .. nat
Frej Jensen
- [Asterisk-Users] AUSTEL Certified
Rainer Jochem
- [Asterisk-Users] AUSTEL Certified
Rainer Jochem
- [Asterisk-Users] MSN Messenger 4.7 vs 5.0
Rainer Jochem
- [Asterisk-Users] MSN Messenger 4.7 vs 5.0
Rainer Jochem
- [Asterisk-Users] grandstream sip phone
Rainer Jochem
- [Asterisk-Users] AVM Fritz! to connect LAN with ISDN line?
Rainer Jochem
- [Asterisk-Users] calls per second with asterisk
Rainer Jochem
- [Asterisk-Users] Cisco 802.11b VoIP phone?
Rainer Jochem
- [Asterisk-Users] Asterisk takes over
Stefan Johnson
- [Asterisk-Users] Asterisk takes over (no answer, but a question)
Stefan Johnson
- [Asterisk-Users] Asterisk takes over
Stefan Johnson
- [Asterisk-Users] (no subject)
Stefan Johnson
- [Asterisk-Users] MSN Messenger 4.7 vs 5.0
Stefan Johnson
- [Asterisk-Users] Asterisk Compile error: make: *** [subdirs] Error 1
Ashley Jones
- [Asterisk-Users] Asterisk Compile error: make: *** [subdirs] Error 1
Ashley Jones
- [Asterisk-Users] ATA-186 de-register
Daryl Jones
- [Asterisk-Users] E&M DID config question
Daryl Jones
- [Asterisk-Users] E&M DID config question
Daryl Jones
- [Asterisk-Users] Robbed bit signalling debugging
Daryl Jones
- [Asterisk-Users] Using asterisk for a 911 call center....
Daryl Jones
- [Asterisk-Users] voicemail enhancements
Daryl Jones
- [Asterisk-Users] Offering an Asterisk Documentation and FAQ Portal
Senad Jordanovic
- [Asterisk-Users] Offering an Asterisk Documentation and FAQ Portal
Senad Jordanovic
- [Asterisk-Users] FWD trouble - 407 error
James H. Cloos Jr.
- [Asterisk-Users] Virtual fax on the Asterisk box
James H. Cloos Jr.
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
James H. Cloos Jr.
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
James H. Cloos Jr.
- [Asterisk-Users] Dynamically setting up/tearing down extensions
James H. Cloos Jr.
- [Asterisk-Users] Ideal Prompt Recording Setup?
James H. Cloos Jr.
- [Asterisk-Users] SIP info
James H. Cloos Jr.
- [Asterisk-Users] the 'pound' and '#' are the same?
James H. Cloos Jr.
- [Asterisk-Users] Conference calls
Klaus-Peter Junghanns
- [Asterisk-Users] *--IAX--* problems. (chan_capi problem)
Klaus-Peter Junghanns
- [Asterisk-Users] Asterisk IVR and Hicom 300
Klaus-Peter Junghanns
- [Asterisk-Users] Configuration sample for isdn4linux?
Klaus-Peter Junghanns
- [Asterisk-Users] executing an agi script after a successful Dial
Klaus-Peter Junghanns
- AW: [Asterisk-Users] * behind ISDN pbx - Forwarding to
extensions with in primary pbx
Klaus-Peter Junghanns
- [Asterisk-Users] Problems with two B channels
Klaus-Peter Junghanns
- [Asterisk-Users] Problems with two B channels
Klaus-Peter Junghanns
- [Asterisk-Users] CAPI & CLID
Klaus-Peter Junghanns
- [Asterisk-Users] SCO/Linux concerns
Ajit M Kallingal
- [Asterisk-Users] A solution for SIP and NAT
Michael Kane
- [Asterisk-Users] A solution for SIP and NAT
Michael Kane
- [Asterisk-Users] A solution for SIP and NAT
Michael Kane
- [Asterisk-Users] A solution for SIP and NAT
Michael Kane
- [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite
Michael Kane
- [Asterisk-Users] A solution for SIP and NAT
Michael Kane
- [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite
Michael Kane
- [Asterisk-Users] A solution for SIP and NAT
Michael Kane
- [Asterisk-Users] Sip call dropping
Michael Kane
- [Asterisk-Users] A solution for SIP and NAT
Michael Kane
- [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite
Michael Kane
- [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite
Michael Kane
- [Asterisk-Users] Crossover T1 cable
Michael Kane
- [Asterisk-Users] Newbie Doubts
Michael Kane
- [Asterisk-Users] Network design question
Michael Kane
- [Asterisk-Users] Connecting two Asterisk servers togerher
Johanna Kangas
- [Asterisk-Users] Contact header empty in SIP-message
Johanna Kangas
- [Asterisk-Users] Getting 488's between two c7940's
Anton L. Kapela
- [Asterisk-Users] Cisco 7960
Anton L. Kapela
- [Asterisk-Users] Activate MySQL logging
Roy Sigurd Karlsbakk
- [Asterisk-Users] Chan_whatever and hanging channels
Roy Sigurd Karlsbakk
- [Asterisk-Users] more abou msn
Roy Sigurd Karlsbakk
- [Asterisk-Users] module : cdr_sybase.so
Roy Sigurd Karlsbakk
- [Asterisk-Users] Asterisk and VMWare
Roy Sigurd Karlsbakk
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Roy Sigurd Karlsbakk
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
Roy Sigurd Karlsbakk
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
Roy Sigurd Karlsbakk
- [Asterisk-Users] Best software SIP client
Roy Sigurd Karlsbakk
- [Asterisk-Users] chan_capi error
Roy Sigurd Karlsbakk
- [Asterisk-Users] MSN Messenger(4.7) Setup
Roy Sigurd Karlsbakk
- [Asterisk-Users] Voicemail() problems - Long pause
afterincoming message recording ended.
Roy Sigurd Karlsbakk
- [Asterisk-Users] can't compile asterisk
Roy Sigurd Karlsbakk
- [Asterisk-Users] time and date stamp in voicemail
Roy Sigurd Karlsbakk
- [Asterisk-Users] VoiceMail2 Wish List
Roy Sigurd Karlsbakk
- [Asterisk-Users] [Solved] CAPI with hanging channels
Roy Sigurd Karlsbakk
- [Asterisk-Users] Asterisk installation
Roy Sigurd Karlsbakk
- [Asterisk-Users] Asterisk installation
Roy Sigurd Karlsbakk
- [Asterisk-Users] Asterisk installation
Roy Sigurd Karlsbakk
- [Asterisk-Users] Microsoft SQL
Roy Sigurd Karlsbakk
- [Asterisk-Users] Some stats
Roy Sigurd Karlsbakk
- [Asterisk-Users] Manager
Roy Sigurd Karlsbakk
- [Asterisk-Users] Manager
Roy Sigurd Karlsbakk
- [Asterisk-Users] (no subject)
Roy Sigurd Karlsbakk
- [Asterisk-Users] Vonage
Roy Sigurd Karlsbakk
- [Asterisk-Users] isdn4linux/Teles16.3
Roy Sigurd Karlsbakk
- [Asterisk-Users] Newbie question
Andrey Katkov
- [Asterisk-Users] Newbie question
Andrey Katkov
- [Asterisk-Users] Re: Newbie question
Andrey Katkov
- [Asterisk-Users] (no subject)
Andrey Katkov
- [Asterisk-Users] dst number
Steven Kawuma
- [Asterisk-Users] dst number
Steven Kawuma
- [Asterisk-Users] Call Accounting
Erik Kendall
- [Asterisk-Users] Call Accounting
Erik Kendall
- [Asterisk-Users] Call Accounting
Erik Kendall
- [Asterisk-Users] Call Accounting
Erik Kendall
- [Asterisk-Users] Call Recording
Erik Kendall
- [Asterisk-Users] Call Recording
Erik Kendall
- [Asterisk-Users] Sip call dropping
Kevin
- [Asterisk-Users] Sip call dropping
Kevin
- [Asterisk-Users] Asterisk IVR and Hicom 300
Maciej Kietlinski
- [Asterisk-Users] Compile Problems with gcc 3.3
Uwe Klein
- [Asterisk-Users] Compile Problems with gcc 3.3
Uwe Klein
- [Asterisk-Users] Hardware support for TDM
Claude Klimos
- [Asterisk-Users] Phones
Nick Knight
- [Asterisk-Users] isdn4linux
Nick Knight
- [Asterisk-Users] Using asterisk for a 911 call center....
Gene Kochanowsky
- [Asterisk-Users] Using asterisk for a 911 call center....
Gene Kochanowsky
- [Asterisk-Users] newbie - simple dialout server
Ferenc Kubinszky
- [Asterisk-Users] Best software SIP client
Erik Lagerway
- [Asterisk-Users] X-Lite and Call transfer using Asterisk
Erik Lagerway
- [Asterisk-Users] X-Lite and Call transfer using Asterisk
Erik Lagerway
- [Asterisk-Users] RFC2833 problems with X-Lite
Erik Lagerway
- [Asterisk-Users] OpenBSD version???
Scott Lambert
- [Asterisk-Users] Phone System Questions
Scott Lambert
- [Asterisk-Users] Techfone VOIP phone
Scott Lambert
- [Asterisk-Users] Cisco 7960g
Siggi Langauf
- [Asterisk-Users] Cisco 7960g
Siggi Langauf
- [Asterisk-Users] Cisco's CallManager and * (was: Cisco 7960g)
Siggi Langauf
- [Asterisk-Users] 2 B channels for ISDN cards
Siggi Langauf
- [Asterisk-Users] Cisco's CallManager and * (was: Cisco 7960g)
(fwd)
Siggi Langauf
- [Asterisk-Users] Asterisk as a stand alone voice mail server
Siggi Langauf
- [Asterisk-Users] Asterisk as a stand alone voice mail server
Siggi Langauf
- [Asterisk-Users] Asterisk as a stand alone voice mail server
Siggi Langauf
- [Asterisk-Users] Cisco's CallManager and * (was: Cisco 7960g)
(fwd)
Siggi Langauf
- [Asterisk-Users] Asterisk as a stand alone voice mail server
Siggi Langauf
- [Asterisk-Users] Cisco 7960g
Siggi Langauf
- [Asterisk-Users] app_voicemail2 became a bit silent, lately...
Siggi Langauf
- [Asterisk-Users] VoiceMail2 Wish List
Siggi Langauf
- [Asterisk-Users] SetCIDName
Siggi Langauf
- [Asterisk-Users] Getting Started with Digium T100/E100 cards
Langley, Sean
- [Asterisk-Users] Asterisk and VMWare
Robert Hajime Lanning
- [Asterisk-Users] conference problem without zapata interface
Robert Hajime Lanning
- [Asterisk-Users] AVM Fritz! to connect LAN with ISDN line?
Achim J. Latz
- [Asterisk-Users] app_festival not cleaning up properly?
John Laur
- [Asterisk-Users] Asterisk and VMWare
John Laur
- [Asterisk-Users] Configuring BudgeTone and ringer over TFTP
John Laur
- [Asterisk-Users] Line Override Device
John Laur
- [Asterisk-Users] module : cdr_sybase.so
John Laur
- [Asterisk-Users] Asterisk and VMWare
John Laur
- [Asterisk-Users] Using 2 PhoneJacks with Asterisk for Data calls.
John Laur
- [Asterisk-Users] Asterisk on Cygwin?
John Laur
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality speech synthesis?
John Laur
- [Asterisk-Users] Multiple Phones for 1 Extension
John Laur
- [Asterisk-Users] Problems getting 7960's to play nice with Asterisk
John Laur
- [Asterisk-Users] Example: Writing a click-to-call application using pbx_spool
John Laur
- [Asterisk-Users] OT: list format vs newsgroup format
John Laur
- [Asterisk-Users] AGI.pm?
John Laur
- [Asterisk-Users] the 'pound' and '#' are the same? (OT Rambling)
John Laur
- [Asterisk-Users] Changes to reset method for ATA186?
John Laur
- [Asterisk-Users] Ordering digital trunks?
John Laur
- [Asterisk-Users] Ordering digital trunks?
John Laur
- [Asterisk-Users] Microsoft SQL
John Laur
- [Asterisk-Users] X100P call detection
Leandro
- [Asterisk-Users] gotoiftime error
Tilghman Lesher
- [Asterisk-Users] gotoiftime error
Tilghman Lesher
- [Asterisk-Users] BIG problem with multiple rings before pickup
Tilghman Lesher
- [Asterisk-Users] BIG problem with multiple rings before pickup
Tilghman Lesher
- [Asterisk-Users] BIG problem with multiple rings before pickup
Tilghman Lesher
- [Asterisk-Users] BIG problem with multiple rings before pickup
Tilghman Lesher
- [Asterisk-Users] System command..
Tilghman Lesher
- [Asterisk-Users] How to modify dialed number?
Tilghman Lesher
- [Asterisk-Users] Why mp3 (licensing issues) as opposed to Open Source OGG
Tilghman Lesher
- [Asterisk-Users] TDM10B - Dies after a few hours
Tilghman Lesher
- [Asterisk-Users] Question About VOIP
Tilghman Lesher
- [Asterisk-Users] Asterisk and VMWare
Tilghman Lesher
- [Asterisk-Users] Odd output from X100P
Tilghman Lesher
- [Asterisk-Users] MSN Messenger 4.7 vs 5.0
Tilghman Lesher
- [Asterisk-Users] Odd output from X100P
Tilghman Lesher
- [Asterisk-Users] module : cdr_sybase.so
Tilghman Lesher
- [Asterisk-Users] Asterisk NOOB
Tilghman Lesher
- [Asterisk-Users] questions
Tilghman Lesher
- [Asterisk-Users] MYSQL Table Structure
Tilghman Lesher
- [Asterisk-Users] executing an agi script after a successful Dial
Tilghman Lesher
- [Asterisk-Users] time and date stamp in voicemail
Tilghman Lesher
- [Asterisk-Users] time and date stamp in voicemail
Tilghman Lesher
- [Asterisk-Users] time and date stamp in voicemail
Tilghman Lesher
- [Asterisk-Users] time and date stamp in voicemail
Tilghman Lesher
- [Asterisk-Users] Linux flavor?
Tilghman Lesher
- [Asterisk-Users] Asterisk installation
Tilghman Lesher
- [Asterisk-Users] voicemail file access problems
Tilghman Lesher
- [Asterisk-Users] voicemail file access problems
Tilghman Lesher
- [Asterisk-Users] Manager
Tilghman Lesher
- [Asterisk-Users] Manager
Tilghman Lesher
- [Asterisk-Users] * Video changes
Tamas Levente
- [Asterisk-Users] msn authentication
Tamas Levente
- [Asterisk-Users] anyone with X100P & Callerid working outside US ?
Tamas Levente
- [Asterisk-Users] EZ-Install
Todd Lieberman
- [Asterisk-Users] Cisco 7960 Transfer & Conference
Todd Lieberman
- [Asterisk-Users] Analog commands
Todd Lieberman
- [Asterisk-Users] how do I do s extensions with PRI
Todd Lieberman
- [Asterisk-Users] VoiceMail2 Wish List
Todd Lieberman
- [Asterisk-Users] VoiceMail2 Wish List
Todd Lieberman
- [Asterisk-Users] voicemail file access problems
Todd Lieberman
- [Asterisk-Users] voicemail file access problems
Todd Lieberman
- [Asterisk-Users] voicemail file access problems
Todd Lieberman
- [Asterisk-Users] cisco5300 with asterisk through H323
George Lin
- [Asterisk-Users] Chan_capi hanging channels
Pavel Litvinenko
- [Asterisk-Users] Cisco 7960
Pavel Litvinenko
- [Asterisk-Users] SetLanguage application doesn;t seem to work in latest Asterisk
William Lloyd
- [Asterisk-Users] 7960 SIP problem when calling from outside of LAN
William Lloyd
- [Asterisk-Users] Silly questions due to ingrained knowledge of analog phone use.
Benjamin Long
- [Asterisk-Users] SIp Registration
Alex Lopez
- [Asterisk-Users] Loaded latest CVS and get Broken PIPE!!!
Alex Lopez
- [Asterisk-Users] RE: IAXTEL toll-free From: Asterisk-Users digest, Vol 1 #791 - 10 msgs
Alex Lopez
- [Asterisk-Users] E911 and asterisk
Alex Lopez
- [Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #873 - 16 msgs
Alex Lopez
- [Asterisk-Users] Re: SIP Authentication bug?
Alex Lopez
- [Asterisk-Users] T410P and zaptel.conf
Alex Lopez
- [Asterisk-Users] INFO: How the T410P sets the number of channels per span
Alex Lopez
- [Asterisk-Users] Following completion when Dialing.
Alex Lopez
- [Asterisk-Users] Asterisk -> AS5300 SIP Interoperability
Low, Adam
- [Asterisk-Users] Asterisk -> AS5300 SIP Interoperability
Low, Adam
- [Asterisk-Users] Cisco 7960
Low, Adam
- [Asterisk-Users] [Asterisk-Users]Help Needed
Low, Adam
- [Asterisk-Users] Help Needed
Low, Adam
- [Asterisk-Users] Asterisk -> SIP -> AS5300 signalling missing on connect/clear cal
l
Low, Adam
- [Asterisk-Users] SIP Call Forwarding/Transfer support ?
Low, Adam
- [Asterisk-Users] SIP Call Forwarding/Transfer support ?
Low, Adam
- [Asterisk-Users] Dialogic hardware
Low, Adam
- [Asterisk-Users] 7940 & AS5300 codec issues/questions G.729 & G.711
Low, Adam
- [Asterisk-Users] can't get musiconhold to work
Low, Adam
- [Asterisk-Users] RTP session traversing Asterisk server ...
Low, Adam
- [Asterisk-Users] RTP session traversing Asterisk server ...
Low, Adam
- [Asterisk-Users] stupid questions ..
Low, Adam
- [Asterisk-Users] stupid questions ..
Low, Adam
- [Asterisk-Users] Linux flavor?
Low, Adam
- [Asterisk-Users] 7960 SIP problem when calling from outside o
f LAN
Low, Adam
- [Asterisk-Users] RTP session traversing Asterisk server ...
Low, Adam
- [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Low, Adam
- [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Low, Adam
- [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Low, Adam
- [Asterisk-Users] Enhanced queue app
Richard Lyman
- [Asterisk-Users] Enhanced queue app
Richard Lyman
- [Asterisk-Users] Problem with digit 0 X-lite
Richard Lyman
- [Asterisk-Users] Problem with SIP Phone with outgoing phone call
John M
- [Asterisk-Users] Voicemail answers, but drops SIP call after about 3 seconds.
Leif Madsen
- [Asterisk-Users] Voicemail answers, but drops SIP call after about 3 seconds.
Leif Madsen
- [Asterisk-Users] Sip CANCEL or BYE when picking up a call ?
Leif Madsen
- [Asterisk-Users] X-Lite Build 1016
Leif Madsen
- [Asterisk-Users] FWD no longer works.. but nothing has changed? Wierd DEBUG errors.
Leif Madsen
- [Asterisk-Users] Best software SIP client
Asterisk Maillist
- [Asterisk-Users] Conference calls
Serge Mankovski
- [Asterisk-Users] SIP call from one extention to another
Serge Mankovski
- [Asterisk-Users] gotoiftime error
Paulo Mannheimer
- [Asterisk-Users] gotoiftime error
Paulo Mannheimer
- [Asterisk-Users] Enhanced queue app
Paulo Mannheimer
- [Asterisk-Users] Enhanced queue app
Paulo Mannheimer
- [Asterisk-Users] problems with new FXS module
Paulo Mannheimer
- [Asterisk-Users] conection with other PBX's
Paulo Mannheimer
- [Asterisk-Users] busydetect and random hangups
Paulo Mannheimer
- [Asterisk-Users] new voicemail messages
Paulo Mannheimer
- [Asterisk-Users] busydetect and random hangups
Paulo Mannheimer
- [Asterisk-Users] Call Dropping
Paulo Mannheimer
- [Asterisk-Users] memory leak in voicemail.c
Paulo Mannheimer
- [Asterisk-Users] voicemail file access problems
Paulo Mannheimer
- [Asterisk-Users] voicemail file access problems
Paulo Mannheimer
- [Asterisk-Users] voicemail file access problems
Paulo Mannheimer
- [Asterisk-Users] random hangups
Paulo H. Mannheimer
- [Asterisk-Users] random hangups
Paulo H. Mannheimer
- [Asterisk-Users] queue bug?
Paulo H. Mannheimer
- [Asterisk-Users] queue bug?
Paulo H. Mannheimer
- [Asterisk-Users] Unable to get SetMusicOnHold working...
Michael Manousos
- [Asterisk-Users] Patch to fix some segfaults in Asterisk
Michael Manousos
- [Asterisk-Users] asterisk-oh323 v0.5.3
Michael Manousos
- [Asterisk-Users] oh323 problem (small one)
Michael Manousos
- [Asterisk-Users] oh323 prob :)
Michael Manousos
- [Asterisk-Users] Matching winth asterisk-oh323
Michael Manousos
- [Asterisk-Users] asterisk-oh323 v0.5.3
Michael Manousos
- [Asterisk-Users] H450 problems
Michael Manousos
- [Asterisk-Users] OH323 + G729 + Go2Call
Michael Manousos
- [Asterisk-Users] OH323 + G729 + Go2Call
Michael Manousos
- [Asterisk-Users] Open H.323 and cdr problem
Michael Manousos
- [Asterisk-Users] Segmentation fault with chan_oh323
Michael Manousos
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Michael Manousos
- [Asterisk-Users] g729 + oh323
Michael Manousos
- [Asterisk-Users] g729 + oh323
Michael Manousos
- [Asterisk-Users] g729 + oh323
Michael Manousos
- [Asterisk-Users] h323 and oh323 modules
Michael Manousos
- [Asterisk-Users] 2 B channels for ISDN cards
Michael Manousos
- [Asterisk-Users] asterisk-oh323 v0.5.4
Michael Manousos
- [Asterisk-Users] h323 and oh323 modules
Michael Manousos
- [Asterisk-Users] h323 gateway call lost after 74sec always
Michael Manousos
- [Asterisk-Users] Debian Package asterisk-oh323?
Michael Manousos
- [Asterisk-Users] h323 gateway call lost after 74sec always
Michael Manousos
- [Asterisk-Users] Problems with two B channels
Michael Manousos
- [Asterisk-Users] Problems with two B channels
Michael Manousos
- [Asterisk-Users] Problems with two B channels
Michael Manousos
- [Asterisk-Users] Re: [Asterisk] help with extension switching
Mario Maqueda
- [Asterisk-Users] * with external sip proxy
Tebaldi Marco
- [Asterisk-Users] mgcp problems
Marsico, Gustavo - (Arg)
- [Asterisk-Users] Asterisk -> AS5300 SIP Interoperability
Marsico, Gustavo - (Arg)
- [Asterisk-Users] Ringing in sequence
Aaron Martin
- [Asterisk-Users] H450 problems
Aaron Martin
- [Asterisk-Users] H450 problems
Aaron Martin
- [Asterisk-Users] H450 problems
Aaron Martin
- [Asterisk-Users] H323 & Transfer
Aaron Martin
- [Asterisk-Users] Call Parking
Aaron Martin
- [Asterisk-Users] Call Parking
Aaron Martin
- [Asterisk-Users] Self parked but avaliable
Aaron Martin
- [Asterisk-Users] MYSQL Table Structure
Aaron Martin
- [Asterisk-Users] MYSQL Table Structure
Aaron Martin
- [Asterisk-Users] Asterisk & X-Lite
Aaron Martin
- [Asterisk-Users] Grandstream Budgettone 100 & 102
Aaron Martin
- [Asterisk-Users] Asterisk PBX Billing
Chris Mason
- [Asterisk-Users] Asterisk PBX Billing
Chris Mason
- [Asterisk-Users] Channel Bank configuration
Marty Mastera
- [Asterisk-Users] Channel Bank configuration
Marty Mastera
- [Asterisk-Users] Channel Bank configuration
Marty Mastera
- [Asterisk-Users] Sip call pickup ?
Brancaleoni Matteo
- [Asterisk-Users] Using switch =>
Brancaleoni Matteo
- [Asterisk-Users] Need a recommendation on a good
motherboard/processor combination
Brancaleoni Matteo
- [Asterisk-Users] No ringing when I dial an extension
Brancaleoni Matteo
- [Asterisk-Users] res parking patch
Brancaleoni Matteo
- [Asterisk-Users] Cllecting digits.
Brancaleoni Matteo
- [Asterisk-Users] PCI Master Abort
Brancaleoni Matteo
- [Asterisk-Users] Call Accounting
Brancaleoni Matteo
- [Asterisk-Users] Budgetone and Voicemail
Brancaleoni Matteo
- [Asterisk-Users] Call Accounting
Brancaleoni Matteo
- [Asterisk-Users] Sip CANCEL or BYE when picking up a call ?
Brancaleoni Matteo
- [Asterisk-Users] Unable to find IP address???
Brancaleoni Matteo
- [Asterisk-Users] Configuring BudgeTone and ringer over TFTP
Brancaleoni Matteo
- [Asterisk-Users] Asterisk and VMWare
Brancaleoni Matteo
- [Asterisk-Users] New budgetone firmware
Brancaleoni Matteo
- [Asterisk-Users] Conditional Contexts
Brancaleoni Matteo
- [Asterisk-Users] Conditional Contexts
Brancaleoni Matteo
- [Asterisk-Users] g723.1 voicemail/conference files segfault *
Brancaleoni Matteo
- [Asterisk-Users] Sip codec preferences
Brancaleoni Matteo
- [Asterisk-Users] IAX pauses
Brancaleoni Matteo
- [Asterisk-Users] IAX pauses
Brancaleoni Matteo
- [Asterisk-Users] busydetect and random hangups
Brancaleoni Matteo
- [Asterisk-Users] busydetect and random hangups
Brancaleoni Matteo
- [Asterisk-Users] SIP info
Brancaleoni Matteo
- [PATCH] [Asterisk-Users] SIP info
Brancaleoni Matteo
- [Asterisk-Users] dialogic drivers
Alastair Maw
- [Asterisk-Users] Asterisk Sacrifice?
Kelly McDonald
- [Asterisk-Users] Asterisk Sacrifice?
Kelly McDonald
- [Asterisk-Users] Re: h323 gateway call lost after 74sec always
Kelly McDonald
- [Asterisk-Users] MGCP behind NAT
Darren McIntosh
- [Asterisk-Users] Australian Options
Mark McKibbin
- [Asterisk-Users] Australian Options
Mark McKibbin
- [Asterisk-Users] Australian Options
Mark McKibbin
- [Asterisk-Users] H.323 Gateway Connection
Jeremy McNamara
- [Asterisk-Users] Fw: SIP Client X-Lite
Jeremy McNamara
- [Asterisk-Users] RTP.C codec error 19
Jeremy McNamara
- [Asterisk-Users] Call Accounting
Jeremy McNamara
- [Asterisk-Users] MGCP-H323v2 transcoder?
Jeremy McNamara
- [Asterisk-Users] SIP call transfers - any other way than using
'#' ?
Jeremy McNamara
- [Asterisk-Users] OH323 + G729 + Go2Call
Jeremy McNamara
- [Asterisk-Users] OH323 + G729 + Go2Call
Jeremy McNamara
- [Asterisk-Users] OH323 + G729 + Go2Call
Jeremy McNamara
- [Asterisk-Users] OH323 + G729 + Go2Call
Jeremy McNamara
- [Asterisk-Users] No Sound via Sip Phone
Jeremy McNamara
- [Asterisk-Users] VIP 30 phone
Jeremy McNamara
- [Asterisk-Users] AUSTEL Certified
Jeremy McNamara
- [Asterisk-Users] Drop the call in 10min
Jeremy McNamara
- [Asterisk-Users] Open H.323 and cdr problem
Jeremy McNamara
- [Asterisk-Users] Using 2 PhoneJacks with Asterisk for Data calls.
Jeremy McNamara
- [Asterisk-Users] Using 2 PhoneJacks with Asterisk for Data calls.
Jeremy McNamara
- [Asterisk-Users] Using 2 PhoneJacks with Asterisk for Data calls.
Jeremy McNamara
- [Asterisk-Users] H323/No one is available to answer at this time
Jeremy McNamara
- [Asterisk-Users] H323/No one is available to answer at this time
Jeremy McNamara
- [Asterisk-Users] VoIP in hotels
Jeremy McNamara
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
Jeremy McNamara
- [Asterisk-Users] chan_h323
Jeremy McNamara
- [Asterisk-Users] Asterisk crashes when trying to load G.729 module.
Jeremy McNamara
- [Asterisk-Users] Best E1 channel bank?
Jeremy McNamara
- [Asterisk-Users] g729 + oh323
Jeremy McNamara
- [Asterisk-Users] Dynamically setting up/tearing down extensions
Jeremy McNamara
- [Asterisk-Users] g729 + oh323
Jeremy McNamara
- [Asterisk-Users] G729 doesn't work.
Jeremy McNamara
- [Asterisk-Users] g729 + oh323
Jeremy McNamara
- [Asterisk-Users] g729 + oh323
Jeremy McNamara
- [Asterisk-Users] h323 and oh323 modules
Jeremy McNamara
- [Asterisk-Users] QoS for Asterisk
Jeremy McNamara
- [Asterisk-Users] h323 and oh323 modules
Jeremy McNamara
- [Asterisk-Users] No callerid on outgoing call over chan_h323
Jeremy McNamara
- [Asterisk-Users] h323 and oh323 modules
Jeremy McNamara
- [Asterisk-Users] No callerid on outgoing call over chan_h323
Jeremy McNamara
- [Asterisk-Users] No callerid on outgoing call over chan_h323
Jeremy McNamara
- [Asterisk-Users] executing an agi script after a successful Dial
Jeremy McNamara
- [Asterisk-Users] Cisco's CallManager and *
Jeremy McNamara
- [Asterisk-Users] fxs without fxo
Jeremy McNamara
- [Asterisk-Users] T410P and zaptel.conf
Jeremy McNamara
- [Asterisk-Users] Asterisk as a stand alone voice mail server
Jeremy McNamara
- [Asterisk-Users] Asterisk as a stand alone voice mail server
Jeremy McNamara
- [Asterisk-Users] Asterisk as a stand alone voice mail server
Jeremy McNamara
- [Asterisk-Users] can't get musiconhold to work
Jeremy McNamara
- [Asterisk-Users] Asterisk user guide ..
Jeremy McNamara
- [Asterisk-Users] SetCIDName
Jeremy McNamara
- [Asterisk-Users] SetCIDName
Jeremy McNamara
- [Asterisk-Users] sample.call + chan_h323 gives seg fault
Jeremy McNamara
- [Asterisk-Users] Vonage
Jeremy McNamara
- [Asterisk-Users] Vonage
Jeremy McNamara
- [Asterisk-Users] 24port or higher fxs
Jeremy McNamara
- [Asterisk-Users] Vonage
Steve Meyers
- [Asterisk-Users] Vonage
Steve Meyers
- [Asterisk-Users] Vonage
Steve Meyers
- [Asterisk-Users] Vonage
Steve Meyers
- [Asterisk-Users] Vonage
Steve Meyers
- [Asterisk-Users] Asterisk - Protocol Converter from SIP/H.323
Sam Michelson
- [Asterisk-Users] Cisco ATA Advanced CallerID
Pauline Middelink
- [Asterisk-Users] Cisco ATA Advanced CallerID
Pauline Middelink
- [Asterisk-Users] Enhanced queue app
Benjamin Miller
- [Asterisk-Users] Voicemail() problems - Long pause after incoming message recording ended.
Benjamin Miller
- [Asterisk-Users] Voicemail message forwarded to another extension and file format changing
Benjamin Miller
- [Asterisk-Users] VoiceMail2 Wish List
Benjamin Miller
- [Asterisk-Users] picking up a ringing extension
Louis-David Mitterrand
- [Asterisk-Users] 7960 SIP problem when calling from outside of LAN
Louis-David Mitterrand
- [Asterisk-Users] Asterisk IVR and Hicom 300
Dr. Andreas Moroder
- [Asterisk-Users] Asterisk IVR and Hicom 300
Dr. Andreas Moroder
- [Asterisk-Users] Integrating cell phone into Asterisk Extension..
John Morris
- [Asterisk-Users] Integrating cell phone into Asterisk Extension..
John Morris
- [Asterisk-Users] Integrating cell phone into Asterisk Extension..
John Morris
- [Asterisk-Users] Integrating cell phone into Asterisk Extension..
John Morris
- [Asterisk-Users] Integrating cell phone into Asterisk Extension..
John Morris
- [Asterisk-Users] Sip: problem authenticating (with Cisco VoIP IOS 12.x) [long]
Simon J Mudd
- [Asterisk-Users] msn
Masakazu Nakano
- [Asterisk-Users] DTMF crashes chan_capi
Jamie Neil
- [Asterisk-Users] DTMF crashes chan_capi
Jamie Neil
- [Asterisk-Users] Ideal Prompt Recording Setup?
Jamie Neil
- [Asterisk-Users] RFC2833 problems with X-Lite
Jamie Neil
- [Asterisk-Users] RFC2833 problems with X-Lite
Jamie Neil
- [Asterisk-Users] RFC2833 problems with X-Lite
Jamie Neil
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality speech synthesis?
Jeff Noxon
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality speech synthesis?
Jeff Noxon
- [Asterisk-Users] Debian Package asterisk-oh323?
Jeff Noxon
- [Asterisk-Users] E1 R2 on Asterisk
Alejandro Olchik
- [Asterisk-Users] res parking patch
Florian Overkamp
- [Asterisk-Users] Direct entry to your own voice mailbox
Florian Overkamp
- [Asterisk-Users] msn
Florian Overkamp
- [Asterisk-Users] hardware requirements
Florian Overkamp
- [Asterisk-Users] VIP 30 phone
Florian Overkamp
- [Asterisk-Users] VIP 30 phone
Florian Overkamp
- [Asterisk-Users] Poll - Would you pay $30-$50 for high
quality speech synthesis?
Florian Overkamp
- [Asterisk-Users] Poll - Would you pay $30-$50 for high
quality speech synthesis?
Florian Overkamp
- [Asterisk-Users] voicemail instructions
Florian Overkamp
- [Asterisk-Users] voicemail instructions
Florian Overkamp
- [Asterisk-Users] Microsoft SQL
Florian Overkamp
- [Asterisk-Users] Manager.pm port
Florian Overkamp
- [Asterisk-Users] Manager.pm port
Florian Overkamp
- [Asterisk-Users] Problem with echo
Dave Packham
- [Asterisk-Users] Linejack strikes again.
Dave Packham
- [Asterisk-Users] SIP canreinvite=yes Broke?
Dave Packham
- [Asterisk-Users] Video Phones?
Dave Packham
- [Asterisk-Users] Asterisk as a stand alone voice mail
server
Dave Packham
- [Asterisk-Users] Best software SIP client
Dave Packham
- [Asterisk-Users] Best software SIP client
Dave Packham
- [Asterisk-Users] Web conf files
Dave Packham
- [Asterisk-Users] executing an agi script after a
successful Dial
Dave Packham
- [Asterisk-Users] RTP session traversing Asterisk server
...
Dave Packham
- [Asterisk-Users] RTP session traversing Asterisk server
...
Dave Packham
- [Asterisk-Users] stutter tone for voicemail on SIP
Dave Packham
- [Asterisk-Users] RTP session traversing Asterisk server
...
Dave Packham
- [Asterisk-Users] RTP session traversing Asterisk server
...
Dave Packham
- [Asterisk-Users] TDM400P card only
Matthew Pallotta
- [Asterisk-Users] fxs without fxo
Matthew Pallotta
- [Asterisk-Users] fxs without fxo
Matthew Pallotta
- [Asterisk-Users] fxs without fxo
Matthew Pallotta
- [Asterisk-Users] A solution for SIP and NAT
Patrick
- [Asterisk-Users] Asterisk and VMWare
Patrick
- [Asterisk-Users] grandstream sip phone
Patrick
- [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Patrick
- [Asterisk-Users] sip -> h323 -> ptsn
Patrick
- [Asterisk-Users] voicemail file access problems
Patrick
- [Asterisk-Users] sip -> h323 -> ptsn
Patrick
- [Asterisk-Users] sip -> h323 -> ptsn
Patrick
- [Asterisk-Users] voicemail file access problems
Patrick
- [Asterisk-Users] unresolved symbols in /lib/modules/2.4.18/misc/zaptel.o
Tim Petlock
- [Asterisk-Users] re: unresolved symbols in /lib/modules/2.4.18/misc/zaptel.o
Tim Petlock
- [Asterisk-Users] can't compile asterisk
Tim Petlock
- [Asterisk-Users] Voicetronix Hardware
Phil
- [Asterisk-Users] Need help
Donn W. Pike
- [Asterisk-Users] T1 config for robbed-bit E&M AMI
Don Pobanz
- [Asterisk-Users] channel bank configuration
Don Pobanz
- [Asterisk-Users] interfacing asterisk with a legacy PBX
Don Pobanz
- [Asterisk-Users] extensions
Don Pobanz
- [Asterisk-Users] Nortel 350
Don Pobanz
- [Asterisk-Users] Nortel 350
Don Pobanz
- [Asterisk-Users] Nortel 350
Don Pobanz
- [Asterisk-Users] VIP 30 phone
Darren Poulson
- [Asterisk-Users] VIP 30 phone
Darren Poulson
- [Asterisk-Users] VIP 30 phone
Darren Poulson
- [Asterisk-Users] Asterisk NOOB
Darren Poulson
- [Asterisk-Users] Analog phone not ringing
Darren Poulson
- [Asterisk-Users] line battery check
Jon Pounder
- [Asterisk-Users] line battery check
Jon Pounder
- [Asterisk-Users] It's true - Nikotel charge for
not-completed calls
Jon Pounder
- [Asterisk-Users] Asterisk Sacrifice?
Andy Powell
- [Asterisk-Users] switch => priority in the dialplan..
(probably an issue for Mark)
Andy Powell
- [Asterisk-Users] Zaptel Alarms/Manager interface
Andy Powell
- [Asterisk-Users] Asterisk and VMWare
Andy Powell
- [Asterisk-Users] Call Transfer
Andy Powell
- [Asterisk-Users] Dialogic hardware
Marcel Prisi
- [Asterisk-Users] Best software SIP client
Marcel Prisi
- [Asterisk-Users] modules.conf again
Emanuele Pucciarelli
- [Asterisk-Users] Remote Agents
Emanuele Pucciarelli
- [Asterisk-Users] *--IAX--* problems. (chan_capi problem)
Emanuele Pucciarelli
- [Asterisk-Users] capi_chan error - CAPI not loaded.
Emanuele Pucciarelli
- [Asterisk-Users] BIG problem with multiple rings before pickup
Karl Putland
- [Asterisk-Users] switch => priority in the dialplan..
(probably an issue for Mark)
Karl Putland
- [Asterisk-Users] mgcp problems
Karl Putland
- [Asterisk-Users] mgcp problems
Karl Putland
- [Asterisk-Users] Hook Flash INFO messages
Karl Putland
- [Asterisk-Users] Conference calls
Martin Pycko
- [Asterisk-Users] picking up a ringing extension
Martin Pycko
- [Asterisk-Users] Unable to get SetMusicOnHold working...
Martin Pycko
- [Asterisk-Users] How do i make best use of Macro?
Martin Pycko
- [Asterisk-Users] *8 pickup then transfer drops call
Martin Pycko
- [Asterisk-Users] BIG problem with multiple rings before pickup
Martin Pycko
- [Asterisk-Users] client reinvitation problem
Martin Pycko
- [Asterisk-Users] res parking patch
Martin Pycko
- [Asterisk-Users] dst number
Martin Pycko
- [Asterisk-Users] Accounting info for SIP Calls
Martin Pycko
- [Asterisk-Users] LD accontability
Martin Pycko
- [Asterisk-Users] PCI Master abort (debian / S400P & X101P
Martin Pycko
- [Asterisk-Users] zt_pri_errors: PRI got event: 8 / 6
Martin Pycko
- [Asterisk-Users] Newbie Doubts
Martin Pycko
- [Asterisk-Users] System command..
Martin Pycko
- [Asterisk-Users] overlap dialing on a pri span
Martin Pycko
- [Asterisk-Users] Can't access outside voicemail services through
asterisk
Martin Pycko
- [Asterisk-Users] Transfert call
Martin Pycko
- [Asterisk-Users] overlap dialing on a pri span
Martin Pycko
- [Asterisk-Users] Using multiple iconnecthere accounts
Martin Pycko
- [Asterisk-Users] Using multiple iconnecthere accounts
Martin Pycko
- [Asterisk-Users] Using multiple iconnecthere accounts
Martin Pycko
- [Asterisk-Users] Using multiple iconnecthere accounts
Martin Pycko
- [Asterisk-Users] callerid= being ignored
Martin Pycko
- [Asterisk-Users] PRI with variable length numbers
Martin Pycko
- [Asterisk-Users] How to modify dialed number?
Martin Pycko
- [Asterisk-Users] How to modify dialed number?
Martin Pycko
- [Asterisk-Users] IAX2 Warning
Martin Pycko
- [Asterisk-Users] T1 config for robbed-bit E&M AMI
Martin Pycko
- [Asterisk-Users] ISDN PRI E1 configuration with E100P
Martin Pycko
- [Asterisk-Users] ISDN PRI E1 configuration with E100P
Martin Pycko
- [Asterisk-Users] Call Recording
Martin Pycko
- [Asterisk-Users] Making Analog Phones Work
Martin Pycko
- [Asterisk-Users] Chan_H323, G729 (minor problem)
Martin Pycko
- [Asterisk-Users] Re: Alphanumerical digits
Martin Pycko
- [Asterisk-Users] Call Pickup
Martin Pycko
- [Asterisk-Users] random hangups
Martin Pycko
- [Asterisk-Users] Chan_H323, G729 (minor problem)
Martin Pycko
- [Asterisk-Users] Asterisk NOOB
Martin Pycko
- [Asterisk-Users] Asterisk crashes when trying to load G.729
module.
Martin Pycko
- [Asterisk-Users] E911 and asterisk
Martin Pycko
- [Asterisk-Users] anyone with X100P & Callerid working outside US ?
Martin Pycko
- [Asterisk-Users] anyone with X100P & Callerid working outside
US ?
Martin Pycko
- [Asterisk-Users] Dynamically setting up/tearing down extensions
Martin Pycko
- [Asterisk-Users] anyone with X100P & Callerid working outside
US ?
Martin Pycko
- [Asterisk-Users] Dynamically setting up/tearing down extensions
Martin Pycko
- [Asterisk-Users] g729 + oh323
Martin Pycko
- [Asterisk-Users] enabling dtmf detection on zap channel?
Martin Pycko
- [Asterisk-Users] busydetect and random hangups
Martin Pycko
- [Asterisk-Users] Asterisk Compile error: make: *** [subdirs]
Error 1
Martin Pycko
- [Asterisk-Users] Problems with g729
Martin Pycko
- [Asterisk-Users] Adtran TSU 600E
Martin Pycko
- [Asterisk-Users] Instant hangup on busy Zap channel.
Martin Pycko
- [Asterisk-Users] TE410P startup
Martin Pycko
- [Asterisk-Users] 'System' application exit with error even if
it performs the job as expected
Martin Pycko
- [Asterisk-Users] 'System' application exit with error even if
it performs the job as expected
Martin Pycko
- [Asterisk-Users] 'System' application exit with error even if
it performs the job as expected - partially solved
Martin Pycko
- [Asterisk-Users] SIP Registration
Martin Pycko
- [Asterisk-Users] Offering an Asterisk Documentation and FAQ Portal
DOUG REISINGER
- [Asterisk-Users] A solution for SIP and NAT
Andrew Radke
- [Asterisk-Users] A solution for SIP and NAT
Andrew Radke
- [Asterisk-Users] A solution for SIP and NAT
Andrew Radke
- [Asterisk-Users] A solution for SIP and NAT
Andrew Radke
- [Asterisk-Users] IVR problem from PSTN phone
Andrzej Radke
- [Asterisk-Users] conference problem without zapata interface
Andrzej Radke
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality speech synthesis?
Greg Renouf
- [Asterisk-Users] grandstream sip phone
Greg Renouf
- [Asterisk-Users] grandstream sip phone
Greg Renouf
- [Asterisk-Users] Manager
Greg Renouf
- [Asterisk-Users] BIG problem with multiple rings before pickup
Sergio Serrano Revuelto
- [Asterisk-Users] .gsm voice format
Sergio Serrano Revuelto
- [Asterisk-Users] Open H.323 and cdr problem
Sergio Serrano Revuelto
- [Asterisk-Users] Open H.323 and cdr problem
Sergio Serrano Revuelto
- [Asterisk-Users] Dial SessionTime
Sergio Serrano Revuelto
- [Asterisk-Users] Two Question
Sergio Serrano Revuelto
- [Asterisk-Users] Two Question
Sergio Serrano Revuelto
- [Asterisk-Users] CDR question
Sergio Serrano Revuelto
- [Asterisk-Users] chan_capi and poor voice quality
Sergio Serrano Revuelto
- [Asterisk-Users] Problem with digit 0 X-lite
Eris Riswanto
- [Asterisk-Users] Problem with digit 0 X-lite
Eris Riswanto
- [Asterisk-Users] Conference calls
Sean P. Robertson
- [Asterisk-Users] Hook Flash INFO messages
Sean P. Robertson
- [Asterisk-Users] Linux flavor?
Sean Rodger
- [Asterisk-Users] Dlink dg102s and G.729
Alexandre Rosa
- [Asterisk-Users] how to set specific codec ?
Alexandre Rosa
- RES: [Asterisk-Users] how to set specific codec ?
Alexandre Rosa
- [Asterisk-Users] D-link 102s and g723 parameters
Alexandre Rosa
- [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Brenton D. Rothchild
- [Asterisk-Users] Problem in compiling g723.1 support in asterisk
Sip Rtp
- [Asterisk-Users] Call Transfer
Sip Rtp
- [Asterisk-Users] Call Transfer
Sip Rtp
- [Asterisk-Users] audio pause/delay problems
Jan Rychter
- [Asterisk-Users] audio pause/delay problems
Jan Rychter
- [Asterisk-Users] audio pause/delay problems
Jan Rychter
- [Asterisk-Users] audio pause/delay problems
Jan Rychter
- [Asterisk-Users] G729 licensing
Jan Rychter
- [Asterisk-Users] audio pause/delay problems
Jan Rychter
- [Asterisk-Users] G729 licensing
Jan Rychter
- [Asterisk-Users] Using 2 PhoneJacks with Asterisk for Data
calls.
Jan Rychter
- [Asterisk-Users] G729 licensing
Jan Rychter
- [Asterisk-Users] audio pause/delay problems
Jan Rychter
- [Asterisk-Users] G729 quality
Jan Rychter
- [Asterisk-Users] IAX pauses
Jan Rychter
- [Asterisk-Users] IAX pauses
Jan Rychter
- [Asterisk-Users] Speex support
Jan Rychter
- [Asterisk-Users] Speex support
Jan Rychter
- [Asterisk-Users] OT: list format vs newsgroup format
Jan Rychter
- [Asterisk-Users] Speex support
Jan Rychter
- [Asterisk-Users] ZapRas: Anyone has an idea how to configure it
???
Thilo Salmon
- [Asterisk-Users] overlap dialing on a pri span
Thilo Salmon
- [Asterisk-Users] hardware requirements
Thilo Salmon
- [Asterisk-Users] enabling dtmf detection on zap channel?
Thilo Salmon
- [Asterisk-Users] * as a softswitch for pri interfaces
Thilo Salmon
- [Asterisk-Users] Busy detect on pri channel?
Thilo Salmon
- [Asterisk-Users] Busy detect on pri channel?
Thilo Salmon
- [Asterisk-Users] retrieving dialed number when overlap dialing?
Thilo Salmon
- [Asterisk-Users] "Forbidden" problem!!
Angelo Sampietro
- [Asterisk-Users] Asterisk PBX Billing
Angelo Sampietro
- [Asterisk-Users] Asterisk PBX Billing
Angelo Sampietro
- [Asterisk-Users] Problem with digit 0 X-lite
Angelo Sampietro
- [Asterisk-Users] asterisk and modem
Angelo Sampietro
- [Asterisk-Users] unsubsribe
Angelo Sampietro
- [Asterisk-Users] *8 pickup then transfer drops call
Chad Sawyer
- [Asterisk-Users] PBX / Asterisk integration
Maik Schmitt
- [Asterisk-Users] IAX2 Warning
Richard Scobie
- [Asterisk-Users] Instant hangup on busy Zap channel.
Richard Scobie
- [Asterisk-Users] Instant hangup on busy Zap channel.
Richard Scobie
- [Asterisk-Users] IAXTel Connect Problem - Mini Frame
Simon Scotland
- [Asterisk-Users] experience with multi-port SIP/FXS gateways?
John Sellens
- [Asterisk-Users] AGI & Silence detection
Nick Seraphin
- [Asterisk-Users] Asterisk as a stand alone voice mail server
Troy Settle
- [Asterisk-Users] Asterisk as a stand alone voice mail server
Troy Settle
- [Asterisk-Users] Asterisk user guide ..
Troy Settle
- [Asterisk-Users] Linux flavor?
Troy Settle
- [Asterisk-Users] BSD (WAS: Linux flavor?)
Troy Settle
- [Asterisk-Users] PHP API for Manager - Plaintext auth needed?
Troy Settle
- [Asterisk-Users] R2 signaling - E1
Artur C. Severo
- [Asterisk-Users] insmod wcfxo failed ( b8zs, esf,
wink startis what I'm trying to do.)
James Sharp
- [Asterisk-Users] BSD (WAS: Linux flavor?)
James Sharp
- [Asterisk-Users] MGCP-H323v2 transcoder?
Sebastian Sill
- [Asterisk-Users] MGCP-H323 interoperability
Sebastian Sill
- [Asterisk-Users] sample.call
Herry Sitepu
- [Asterisk-Users] Budgetone and Voicemail
James Sizemore
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
James Sizemore
- [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
James Sizemore
- [Asterisk-Users] Sip call question
Skuse, Phil
- [Asterisk-Users] Integrate Asterisk with Meridian phone syste
m
Skuse, Phil
- [Asterisk-Users] the 'pound' and '#' are the same? (OT Rambli
ng)
Skuse, Phil
- [Asterisk-Users] RE Pingtel Phones
Skuse, Phil
- [Asterisk-Users] RTP codec 13 received - Cisco incompatibilit
y?
Skuse, Phil
- [Asterisk-Users] Grandstream Budgettone 100 & 102
Skuse, Phil
- [Asterisk-Users] Music on hold & Read error on sound device
Darren Smith
- [Asterisk-Users] X100P and incoming Context + CDR?
Darren Smith
- [Asterisk-Users] How do you force Asterisk to use only
specific codecs?
Jared Smith
- [Asterisk-Users] line battery check
Jared Smith
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality
speech synthesis?
Jared Smith
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality
speech synthesis?
Jared Smith
- [Asterisk-Users] CDR Information and Pipes
Richard Smith
- [Asterisk-Users] Dynamically setting up/tearing down
extensions
Steve Sobol
- [Asterisk-Users] Dynamically setting up/tearing down extensions
Steven J. Sobol
- [Asterisk-Users] Dynamically setting up/tearing down extensions
Steven J. Sobol
- [Asterisk-Users] Dynamically setting up/tearing down extensions
Steven J. Sobol
- [Asterisk-Users] AGI.pm?
Steven J. Sobol
- [Asterisk-Users] Best software SIP client
Steven J. Sobol
- [Asterisk-Users] Best software SIP client
Steven J. Sobol
- [Asterisk-Users] Asterisk user guide ..
Steven J. Sobol
- [Asterisk-Users] Microsoft SQL
Steven J. Sobol
- [Asterisk-Users] Microsoft SQL
Steven J. Sobol
- [Asterisk-Users] X-Lite and Call transfer using Asterisk
Steven J. Sobol
- [Asterisk-Users] Manager.pm port
Steven J. Sobol
- [Asterisk-Users] Manager.pm port
Steven J. Sobol
- [Asterisk-Users] Manager
Steven J. Sobol
- [Asterisk-Users] Manager
Steven J. Sobol
- [Asterisk-Users] PHP API for Manager - Plaintext auth needed?
Steven J. Sobol
- [Asterisk-Users] Enhanced queue app
Mark Spencer
- [Asterisk-Users] Enhanced queue app
Mark Spencer
- [Asterisk-Users] CVS fixed
Mark Spencer
- [Asterisk-Users] Cisco ATA-186 config guide for Asterisk
Mark Spencer
- [Asterisk-Users] BIG problem with multiple rings before pickup
Mark Spencer
- [Asterisk-Users] Using switch =>
Mark Spencer
- [Asterisk-Users] gastman and queues
Mark Spencer
- [Asterisk-Users] switch => priority in the dialplan.. (probably
an issue for Mark)
Mark Spencer
- [Asterisk-Users] How to make * send RTCP reports
Mark Spencer
- [Asterisk-Users] Voicemail2 Contexts
Mark Spencer
- [Asterisk-Users] ZapRas: Anyone has an idea how to configure it
???
Mark Spencer
- [Asterisk-Users] Loaded latest CVS and get Broken PIPE!!!
Mark Spencer
- [Asterisk-Users] Agent in new CVS
Mark Spencer
- Attention TDM400P users (was Re: [Asterisk-Users] PCI Master Abort)
Mark Spencer
- [Asterisk-Users] IAXTEL toll-free
Mark Spencer
- [Asterisk-Users] modules.conf again
Mark Spencer
- [Asterisk-Users] Error on PRI channel : Call specified but not
found!
Mark Spencer
- [Asterisk-Users] Transfers on the Cisco 7960
Mark Spencer
- [Asterisk-Users] Hook Flash INFO messages
Mark Spencer
- [Asterisk-Users] AUSTEL Certified
Mark Spencer
- [Asterisk-Users] AUSTEL Certified
Mark Spencer
- [Asterisk-Users] something is wrong with gsm prompts format
Mark Spencer
- [Asterisk-Users] AUSTEL Certified
Mark Spencer
- [Asterisk-Users] voicemail instructions
Mark Spencer
- [Asterisk-Users] TE410P startup (2 boards)
Mark Spencer
- [Asterisk-Users] serious dtmf recognition problem.
Mark Spencer
- [Asterisk-Users] Self parked but avaliable
Mark Spencer
- [Asterisk-Users] 'Echo' - I'm sure a common topic
Mark Spencer
- [Asterisk-Users] SIP Authentication bug?
Mark Spencer
- [Asterisk-Users] Dynamically setting up/tearing down extensions
Mark Spencer
- [Asterisk-Users] busydetect and random hangups
Mark Spencer
- [Asterisk-Users] T410P and zaptel.conf
Mark Spencer
- [Asterisk-Users] Bug Tracker Official Launch
Mark Spencer
- [Asterisk-Users] moh/playback for non-zap interfaces
Mark Spencer
- [Asterisk-Users] TE410P startup
Mark Spencer
- [Asterisk-Users] time and date stamp in voicemail
Mark Spencer
- [Asterisk-Users] Channel Language
Mark Spencer
- [Asterisk-Users] go on in current context after destination
channels hung up ?
Mark Spencer
- [Asterisk-Users] Offering an Asterisk Documentation and FAQ
Portal
Mark Spencer
- [Asterisk-Users] Offering an Asterisk Documentation and FAQ
Portal
Mark Spencer
- [Asterisk-Users] Following completion when Dialing.
Mark Spencer
- [Asterisk-Users] Variable Substitution
Mark Spencer
- [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Mark Spencer
- [Asterisk-Users] Congestion
Mark Spencer
- [Asterisk-Users] RTP codec 13 received - Cisco incompatibility?
Mark Spencer
- [Asterisk-Users] Help with ON-Hold, and call-transfer.
Mark Spencer
- [Asterisk-Users] SIP calls cause Segmentation Fault
Mark Spencer
- [Asterisk-Users] Vonage
Mark Spencer
- [Asterisk-Users] experience with multi-port SIP/FXS gateways?
Mark Spowage
- [Asterisk-Users] G729 licensing
Mark Spowage
- [Asterisk-Users] Phoneserve SIP provider
Sergey S. Stasyuk
- [Asterisk-Users] Phoneserve SIP provider
Sergey S. Stasyuk
- [Asterisk-Users] TDM400P noise?
Steve
- [Asterisk-Users] time and date stamp in voicemail
Steve
- [Asterisk-Users] SCO/Linux concerns
Steve
- [Asterisk-Users] More mec3 feedback
Iain Stevenson
- [Asterisk-Users] Problem with echo
Iain Stevenson
- [Asterisk-Users] Newbie question
Iain Stevenson
- [Asterisk-Users] FWD trouble - 407 error
Iain Stevenson
- [Asterisk-Users] FWD trouble - 407 error
Iain Stevenson
- [Asterisk-Users] SIP call transfers - any other way than using '#' ?
Iain Stevenson
- [Asterisk-Users] wait and user input..
Iain Stevenson
- [Asterisk-Users] Cisco 7905G vs ATA186
Iain Stevenson
- [Asterisk-Users] Analog phone not ringing
Iain Stevenson
- [Asterisk-Users] FWD no longer works.. but nothing has changed?
Wierd DEBUG errors.
Iain Stevenson
- [Asterisk-Users] ISDN Fritz & RedHat 8.0
Iain Stevenson
- [Asterisk-Users] moh/playback for non-zap interfaces
Iain Stevenson
- [Asterisk-Users] Call transfer on ATA186
Iain Stevenson
- [Asterisk-Users] RTP codec 13 received - Cisco
incompatibilit y?
Iain Stevenson
- [Asterisk-Users] Asterisk PBX Billing
Scott Stingel
- [Asterisk-Users] Is there any real asterisk documentation ?
Scott Stingel
- [Asterisk-Users] line battery check
Scott Stingel
- [Asterisk-Users] .gsm voice format
Scott Stingel
- [Asterisk-Users] .gsm voice format
Scott Stingel
- [Asterisk-Users] .gsm voice format
Scott Stingel
- [Asterisk-Users] Odd output from X100P
Scott Stingel
- [Asterisk-Users] Hardware Vendors
Scott Stingel
- [Asterisk-Users] high quality speech synthesis - AT&T vs the others
Scott Stingel
- [Asterisk-Users] Integrate Asterisk with Meridian phone system
Scott Stingel
- [Asterisk-Users] Self-testing E1 cards?
Scott Stingel
- [Asterisk-Users] the 'pound' and '#' are the same?
Scott Stingel
- [Asterisk-Users] reconnecting
Scott Stingel
- [Asterisk-Users] Problem with AGI "Record File"
Scott Stingel
- [Asterisk-Users] Offering an Asterisk Documentation and FAQ Portal
Scott Stingel
- [Asterisk-Users] Offering an Asterisk Documentation and FAQ Portal
Scott Stingel
- [Asterisk-Users] DID number assignment to SIP phones
Mark Street
- [Asterisk-Users] ZapRas
Lord Stroud
- [Asterisk-Users] ZapRas: Anyone has an idea how to configure it ???
Lord Stroud
- [Asterisk-Users] RTP.C codec error 19
Lord Stroud
- [Asterisk-Users] RTP.C codec error 19
Lord Stroud
- [Asterisk-Users] UK Gateway
Linus Surguy
- [Asterisk-Users] 'Echo' - I'm sure a common topic
Linus Surguy
- [Asterisk-Users] 'Echo' - I'm sure a common topic
Linus Surguy
- [Asterisk-Users] 'Echo' - I'm sure a common topic
Linus Surguy
- [Asterisk-Users] the 'pound' and '#' are the same?
Linus Surguy
- [Asterisk-Users] Outgoing calls through a calling card
John Sutter
- [Asterisk-Users] Enhanced queue app
TC
- [Asterisk-Users] Enhanced queue app
TC
- [Asterisk-Users] Enhanced queue app
TC
- [Asterisk-Users] Crossover T1 cable
TC
- [Asterisk-Users] incoming callerid on FXO
TC
- [Asterisk-Users] Best Analog sets for use w/*
TC
- [Asterisk-Users] OT: list format vs newsgroup format
James Taylor
- [Asterisk-Users] OT: list format vs newsgroup format
James Taylor
- [Asterisk-Users] questions
James Taylor
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
James Taylor
- [Asterisk-Users] E911 and asterisk
James Taylor
- [Asterisk-Users] Verizon, SBC local company?
James Taylor
- [Asterisk-Users] RTP session traversing Asterisk server...
TeleSIP
- [Asterisk-Users] Unable to get SetMusicOnHold working...
Fabrice Tereszkiewicz
- [Asterisk-Users] Unable to get SetMusicOnHold working...
Fabrice Tereszkiewicz
- [Asterisk-Users] record a conversation
Hervé Thibaud
- [Asterisk-Users] record a conversation
Hervé Thibaud
- [Asterisk-Users] [Newbie] SIP via fwd
Hervé Thibaud
- [Asterisk-Users] [Newbie] SIP via fwd
Hervé Thibaud
- [Asterisk-Users] macro-record-cleanup in extensions.conf
Hervé Thibaud
- [Asterisk-Users] macro-record-cleanup in extensions.conf
Hervé Thibaud
- [Asterisk-Users] macro-record-cleanup in extensions.conf
Hervé Thibaud
- [Asterisk-Users] h323 gateway call lost after 74sec always
Steven Thomas
- [Asterisk-Users] h323 gateway call lost after 74sec always
Steven Thomas
- [Asterisk-Users] h323 gateway call lost after 74sec always
Steven Thomas
- [Asterisk-Users] h323 gateway call lost after 74sec always
Steven Thomas
- [Asterisk-Users] Chan_oh323 Dial format / voice latency 4 to 5 secs
Steven Thomas
- [Asterisk-Users] Offering an Asterisk Documentation and FAQ Portal
James H. Thompson
- [Asterisk-Users] Net2Phone SIP
Mark Thompson
- [Asterisk-Users] Segmentation fault with chan_oh323
Mark Thompson
- [Asterisk-Users] Cisco interoperability?
Anton Tinchev
- [Asterisk-Users] Music while waiting for agent to free.
Anton Tinchev
- [Asterisk-Users] Asterisk crashes when trying to load G.729 module.
Anton Tinchev
- [Asterisk-Users] Asterisk crashes when trying to load G.729 module.
Anton Tinchev
- [Asterisk-Users] Best E1 channel bank?
Anton Tinchev
- [Asterisk-Users] Best E1 channel bank?
Anton Tinchev
- [Asterisk-Users] Best E1 channel bank?
Anton Tinchev
- [Asterisk-Users] Dynamically setting up/tearing down extensions
Anton Tinchev
- [Asterisk-Users] QoS for Asterisk
Anton Tinchev
- [Asterisk-Users] ISDN Callout problem
Anton Tinchev
- [Asterisk-Users] audiocodes fxs
Anton Tinchev
- [Asterisk-Users] Large-scale voicemail deployment: any experiences?
John Todd
- [Asterisk-Users] Enhanced queue app
John Todd
- [Asterisk-Users] A solution for SIP and NAT
John Todd
- [Asterisk-Users] A solution for SIP and NAT
John Todd
- [Asterisk-Users] A solution for SIP and NAT
John Todd
- [Asterisk-Users] A solution for SIP and NAT
John Todd
- [Asterisk-Users] Cisco ATA-186 config guide for Asterisk
John Todd
- [Asterisk-Users] A solution for SIP and NAT
John Todd
- [Asterisk-Users] A solution for SIP and NAT
John Todd
- [Asterisk-Users] Coding Time/Date announce on voicemail
John Todd
- [Asterisk-Users] Please help -- Syntax for dialing VoIP
provider
John Todd
- [Asterisk-Users] Please help -- Syntax for dialing VoIP
provider
John Todd
- [Asterisk-Users] Please help -- Syntax for dialing VoIP
provider
John Todd
- [Asterisk-Users] Accurate Billing
John Todd
- [Asterisk-Users] Network design question
John Todd
- [Asterisk-Users] Asterisk basic how-to on O'Reilly's site
John Todd
- [Asterisk-Users] OpenBSD version???
John Todd
- [Asterisk-Users] Asterisk as SIP <-> PSTN gateway
John Todd
- [Asterisk-Users] More voice prompts available now
John Todd
- [Asterisk-Users] audio pause/delay problems
John Todd
- [Asterisk-Users] SIP immediate hangups with latest CVS
John Todd
- [Asterisk-Users] SIP immediate hangups with latest CVS
John Todd
- [Asterisk-Users] unresolved symbols in
/lib/modules/2.4.18/misc/zaptel.o
John Todd
- [Asterisk-Users] audio pause/delay problems
John Todd
- [Asterisk-Users] SIP immediate hangups with latest CVS
John Todd
- [Asterisk-Users] Fwd:[Vocal] Question about Cisco IP hard phones
John Todd
- [Asterisk-Users] R2 signaling - E1
John Todd
- [Asterisk-Users] Poll - Would you pay $30-$50 for high
quality speech synthesis?
John Todd
- [Asterisk-Users] E1 R2 on Asterisk
John Todd
- [Asterisk-Users] outgoing callerid string
John Todd
- [Asterisk-Users] Can I interoperate with public PSTN gateways
?
John Todd
- [Asterisk-Users] Silly questions due to ingrained knowledge
of analog phone use.
John Todd
- [Asterisk-Users] ATA-186 software upgrade 2.16.1 - notes?
John Todd
- [Asterisk-Users] Speex support
John Todd
- [Asterisk-Users] Speex support
John Todd
- [Asterisk-Users] SIP Authentication bug?
John Todd
- [Asterisk-Users] SIP Call Forwarding/Transfer support ?
John Todd
- [Asterisk-Users] Cisco 7960 upgrade from SKINNY load
John Todd
- [Asterisk-Users] executing an agi script after a successful
Dial
John Todd
- [Asterisk-Users] how do I do s extensions with PRI
John Todd
- [Asterisk-Users] Asterisk as a stand alone voice mail server
John Todd
- [Asterisk-Users] Asterisk as a stand alone voice mail server
John Todd
- [Asterisk-Users] Changes to reset method for ATA186?
John Todd
- [Asterisk-Users] time and date stamp in voicemail
John Todd
- [Asterisk-Users] Busy detect on pri channel?
John Todd
- [Asterisk-Users] Re: Panasonic and Asterisk
Jose Ildefonso Camargo Tolosa
- [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #882 - 11 msgs
Jose Ildefonso Camargo Tolosa
- [Asterisk-Users] Re: [Asterisk-Dev] ANNOUNCE: CLASS-like features for Asterisk
The Traveller
- [Asterisk-Users] Seg Fault!!
The Traveller
- [Asterisk-Users] BIG problem with multiple rings before pickup
The Traveller
- [Asterisk-Users] zt_pri_errors: PRI got event: 8 / 6
The Traveller
- [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything
The Traveller
- [Asterisk-Users] Problems with TDM40P
The Traveller
- [Asterisk-Users] Problems with TDM40P
The Traveller
- [Asterisk-Users] Direct entry to your own voice mailbox
The Traveller
- [Asterisk-Users] Direct entry to your own voice mailbox
The Traveller
- [Asterisk-Users] ignorepat doesn't work
The Traveller
- [Asterisk-Users] PRI with variable length numbers
The Traveller
- [Asterisk-Users] PRI with variable length numbers
The Traveller
- [Asterisk-Users] What does "callerid=" in sip.conf do?
The Traveller
- [Asterisk-Users] Analog commands
The Traveller
- [Asterisk-Users] voicemail instructions
The Traveller
- [Asterisk-Users] IAX pauses
The Traveller
- [Asterisk-Users] XS4ALL Gateway now also does FWD
The Traveller
- [Asterisk-Users] No audio in Messenger
The Traveller
- [Asterisk-Users] enabling dtmf detection on zap channel?
The Traveller
- [Asterisk-Users] capi_chan error - CAPI not loaded.
The Traveller
- [Asterisk-Users] MSN Messenger(4.7) Setup
The Traveller
- [Asterisk-Users] can't get musiconhold to work
The Traveller
- [Asterisk-Users] Problems with chan_sip on multi-homed hosts
The Traveller
- [Asterisk-Users] FWD-gateway prefix
The Traveller
- [Asterisk-Users] iax2 and reinvites
The Traveller
- [Asterisk-Users] Linejack strikes again.
Zara Trousk
- [Asterisk-Users] Linejack strikes again.
Zara Trousk
- [Asterisk-Users] Linejack strikes again.
Zara Trousk
- [Asterisk-Users] What does "callerid=" in sip.conf do?
Ryan Tucker
- [Asterisk-Users] Asterisk takes over
Ryan Tucker
- [Asterisk-Users] SIP call from one extention to another
Ryan Tucker
- [Asterisk-Users] SIP info
Ryan Tucker
- [PATCH] [Asterisk-Users] SIP info
Ryan Tucker
- [Asterisk-Users] Cisco 7960 upgrade from SKINNY load
Ryan Tucker
- [Asterisk-Users] the 'pound' and '#' are the same?
Ryan Tucker
- [Asterisk-Users] Supplementing Current phone system
Ben Turner
- [Asterisk-Users] Fw: SIP Client X-Lite
Jay Tyndall
- [Asterisk-Users] IAX Bandwidth Question
Jay Tyndall
- Fw: [Asterisk-Users] IAX Bandwidth Question
Jay Tyndall
- [Asterisk-Users] IAX G729 Codec
Jay Tyndall
- [Asterisk-Users] IAX G729 Codec
Jay Tyndall
- [Asterisk-Users] Setting up A TDM400P
Jay Tyndall
- [Asterisk-Users] Setting up A TDM400P
Jay Tyndall
- [Asterisk-Users] Making Analog Phones Work
Jay Tyndall
- [Asterisk-Users] Analog commands
Jay Tyndall
- [Asterisk-Users] Call Pickup
Jay Tyndall
- [Asterisk-Users] Call Parking
Jay Tyndall
- [Asterisk-Users] Direct Indial with ISDN and Netjet-S
Jay Tyndall
- [Asterisk-Users] Segmentation fault with chan_oh323
Michael Ulitskiy
- [Asterisk-Users] Segmentation fault with chan_oh323
Michael Ulitskiy
- [Asterisk-Users] No callerid on outgoing call over chan_h323
Michael Ulitskiy
- [Asterisk-Users] No callerid on outgoing call over chan_h323
Michael Ulitskiy
- [Asterisk-Users] No callerid on outgoing call over chan_h323
Michael Ulitskiy
- [Asterisk-Users] Call transfer on ATA186
Michael Ulitskiy
- [Asterisk-Users] Call transfer on ATA186
Michael Ulitskiy
- [Asterisk-Users] * Video changes
Steve Underwood
- [Asterisk-Users] AUSTEL Certified
Steve Underwood
- AW: [Asterisk-Users] mod tor2 takes 20-30% from CPU (20-30% System)
Steve Underwood
- [Asterisk-Users] G729 quality
Steve Underwood
- [Asterisk-Users] Text to Speech - Someone needs to do this
Steve Underwood
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality
speech synthesis?
Steve Underwood
- [Asterisk-Users] Text to Speech - Someone needs to do this
Steve Underwood
- [Asterisk-Users] E1 R2 on Asterisk
Steve Underwood
- [Asterisk-Users] E1 R2 on Asterisk
Steve Underwood
- [Asterisk-Users] serious dtmf recognition problem.
Steve Underwood
- [Asterisk-Users] Techfone VOIP phone
Steve Underwood
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Steve Underwood
- [Asterisk-Users] Techfone VOIP phone
Steve Underwood
- [Asterisk-Users] TE410P startup
Steve Underwood
- [Asterisk-Users] TE410P startup
Steve Underwood
- [Asterisk-Users] Channel Language
Reini Urban
- [Asterisk-Users] Supplementing Current phone system
Dylan VanHerpen
- [Asterisk-Users] ADSI and SoftKeys
Jayson Vantuyl
- [Asterisk-Users] Virtual fax on the Asterisk box
Andrea Venturi
- [Asterisk-Users] asterisk and modem
Armand A. Verstappen
- [Asterisk-Users] EZ-Install
Armand A. Verstappen
- [Asterisk-Users] conference problem without zapata interface
Armand A. Verstappen
- [Asterisk-Users] Dynamically setting up/tearing down extensions
Armand A. Verstappen
- [Asterisk-Users] anyone with X100P & Callerid working outside
US ?
Armand A. Verstappen
- [Asterisk-Users] Cisco ATA Advanced CallerID
Armand A. Verstappen
- [Asterisk-Users] Cisco ATA Advanced CallerID
Armand A. Verstappen
- [Asterisk-Users] Channel Language
Armand A. Verstappen
- [Asterisk-Users] Dummy account/extension
Armand A. Verstappen
- [Asterisk-Users] isdn4linux/Teles16.3
Armand A. Verstappen
- [Asterisk-Users] ADSI and SoftKeys
Armand A. Verstappen
- [Asterisk-Users] Dummy account/extension
Armand A. Verstappen
- [Asterisk-Users] %unsuscribe
Armand A. Verstappen
- [Asterisk-Users] voicemail file access problems
Armand A. Verstappen
- [Asterisk-Users] voicemail file access problems
Armand A. Verstappen
- [Asterisk-Users] ADSI and SoftKeys
Armand A. Verstappen
- [Asterisk-Users] ADSI and SoftKeys
Armand A. Verstappen
- [Asterisk-Users] PCM Voice Quality Issue on CVS Version
Ricardo Villa
- [Asterisk-Users] g729 Codec
Ricardo Villa
- [Asterisk-Users] PCM Voice Quality Issue on CVS Version
Ricardo Villa
- [Asterisk-Users] g729 Codec
Ricardo Villa
- [Asterisk-Users] g729 Codec
Ricardo Villa
- [Asterisk-Users] Grandstream Budgettone 100 & 102
Ricardo Villa
- [Asterisk-Users] Vonage
Ricardo Villa
- [Asterisk-Users] Vonage
Ricardo Villa
- [Asterisk-Users] Vonage
Ricardo Villa
- [Asterisk-Users] Vonage
Ricardo Villa
- [Asterisk-Users] unsubscribe
Vladislav
- [Asterisk-Users] Using 2 PhoneJacks with Asterisk for Data calls.
Lee W
- [Asterisk-Users] Using 2 PhoneJacks with Asterisk for Data calls.
Lee W
- [Asterisk-Users] Using 2 PhoneJacks with Asterisk for Data calls.
Lee W
- [Asterisk-Users] compilation error
John WALTER
- FW: [Asterisk-Users] Sip codec preferences
DUSTIN WILDES
- X100P mod or USB relay box, RE: [Asterisk-Users] Line Override
Device
Reed Wade
- X100P mod or USB relay box, RE: [Asterisk-Users] Line
Override Device
Reed Wade
- X100P mod or USB relay box, RE: [Asterisk-Users] Line
Override Device
Reed Wade
- [Asterisk-Users] Grandstream BudgeTone 102 initial experiences
Reed Wade
- [Asterisk-Users] Budgetone and NTP (redux)
Reed Wade
- [Asterisk-Users] Grandstream BudgeTone 102 initial
experiences
Reed Wade
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
Reed Wade
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
Reed Wade
- [Asterisk-Users] Grandstream Budgettone 100 & 102
Reed Wade
- [Asterisk-Users] Migration to Asterisk - Running off of Merlin
Legend system
Dave Weis
- [Asterisk-Users] Migration to Asterisk - Running off of Merlin
Legend system
Dave Weis
- [Asterisk-Users] Virtual fax on the Asterisk box
Dave Weis
- [Asterisk-Users] Integratting * With Database(Newbie)
God Knows Well
- [Asterisk-Users] Integratting * With Database(Newbie)
God Knows Well
- [Asterisk-Users] Asterisk installation
Wen Wen
- [Asterisk-Users] Asterisk installation
Wen Wen
- [Asterisk-Users] Asterisk installation
Wen Wen
- [Asterisk-Users] Migration to Asterisk - Running off of Merlin Legend system
Wade Weppler
- [Asterisk-Users] callgroup and pickupgroup
Wade Weppler
- [Asterisk-Users] Newbie Doubts
Wade Weppler
- [Asterisk-Users] System command..
Wade Weppler
- [Asterisk-Users] MSN Messenger 4.7 vs 5.0
Wade Weppler
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
Wade Weppler
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
Wade Weppler
- [Asterisk-Users] Asterisk NOOB
Wade Weppler
- [Asterisk-Users] Music on hold & Read error on sound device
Wade Weppler
- [Asterisk-Users] Integrate Asterisk with Meridian phone system
Wade Weppler
- [Asterisk-Users] Codecs for use with Cisco 7960 and ATA-186
Wade Weppler
- [Asterisk-Users] Adtran TSU 600E
Wade Weppler
- [Asterisk-Users] Voicemail() problems - Long pause afterincoming message recording ended.
Wade Weppler
- [Asterisk-Users] Nortel 350
Wade Weppler
- [Asterisk-Users] Nortel 350
Wade Weppler
- [Asterisk-Users] MGCP behind NAT
Wade Weppler
- [Asterisk-Users] Vonage
Wade Weppler
- [Asterisk-Users] Parking calls - why doesn't work?
Wade Weppler
- [Asterisk-Users] Festival talks fast...
Brian West
- [Asterisk-Users] time and date stamp in voicemail
Brian West
- [Asterisk-Users] time and date stamp in voicemail
Brian West
- [Asterisk-Users] VoiceMail2 Wish List
Brian West
- [Asterisk-Users] Hunt group examples?
Brian West
- [Asterisk-Users] Call Forwarding and DND conf
Brian West
- [Asterisk-Users] Linux flavor?
Brian West
- [Asterisk-Users] Call Forwarding and DND conf
Brian West
- [Asterisk-Users] Call Transfer, Budgettone 100
Brian West
- [Asterisk-Users] X-Lite and Call transfer using Asterisk
Brian West
- [Asterisk-Users] sip -> h323 -> ptsn
Brian West
- [Asterisk-Users] sip -> h323 -> ptsn
Brian West
- [Asterisk-Users] sip -> h323 -> ptsn
Brian West
- [Asterisk-Users] Grandstream Budgettone 100 & 102
Brian West
- [Asterisk-Users] sip -> h323 -> ptsn
Brian West
- [Asterisk-Users] AddQueueMember and RemoveQueueMember
Brian West
- [Asterisk-Users] AddQueueMember and RemoveQueueMember
Brian West
- [Asterisk-Users] H3500CW recommendation
Howard White
- [Asterisk-Users] Lost in translation (was the 'pound' and '#'
are the same?)
Howard White
- [Asterisk-Users] More switch => stuff
Eric Wieling
- [Asterisk-Users] More switch => stuff
Eric Wieling
- [Asterisk-Users] How do you force Asterisk to use only
specific codecs?
Eric Wieling
- [Asterisk-Users] sip -> h323 -> ptsn
Eric Wieling
- [Asterisk-Users] Configuration sample for isdn4linux?
Holger Wirtz
- [Asterisk-Users] incoming callerid on FXO
Sunny Woo
- [Asterisk-Users] AGI script sample using bash shell script
Sunny Woo
- [Asterisk-Users] AGI script sample using bash shell script
Sunny Woo
- [Asterisk-Users] Asterisk and VMWare
Sunny Woo
- [Asterisk-Users] Digital phones
Anthony Wood
- X100P mod or USB relay box, RE: [Asterisk-Users] Line Override Device
Anthony Wood
- [Asterisk-Users] AUSTEL Certified
Anthony Wood
- [Asterisk-Users] Australian Options
Anthony Wood
- [Asterisk-Users] Asterisk and Hot Desks??
Simon Woodhead
- [Asterisk-Users] Virtual fax on the Asterisk box
Simon Woodhead
- [Asterisk-Users] Virtual fax on the Asterisk box
Simon Woodhead
- [Asterisk-Users] IAX Bandwidth Question
Simon Woodhead
- [Asterisk-Users] IAX G729 Codec
Simon Woodhead
- [Asterisk-Users] IAX G729 Codec
Simon Woodhead
- [Asterisk-Users] IAX G729 Codec
Simon Woodhead
- [Asterisk-Users] Again Asterisk and VMWare - it works now!
Simon Woodhead
- [Asterisk-Users] Phones
Simon Woodhead
- [Asterisk-Users] Best software SIP client
Simon Woodhead
- [Asterisk-Users] SIP Registration
Steve Woolley
- [Asterisk-Users] Problems with musiconhold
Xisco
- [Asterisk-Users] Accurate Billing
Xisco
- [Asterisk-Users] Initiations in IP voice/Hybrid Voice/etc...
Xisco
- [Asterisk-Users] Problem with meetme.
Xisco
- [Asterisk-Users] VoIP in hotels
Jonathan Young
- [Asterisk-Users] Today's Message from linphone; update on Khpone and SJPhone and X-Lite
Moshe Yudkowsky
- [Asterisk-Users] Today's Message from linphone; update on
Khpone and SJPhone and X-Lite
Moshe Yudkowsky
- [Asterisk-Users] Today's Message from linphone; update on
Khpone and SJPhone and X-Lite
Moshe Yudkowsky
- [Asterisk-Users] Sorry 'bout that
Moshe Yudkowsky
- [Asterisk-Users] One-way talk paths (without INVITE?) and other issues
Moshe Yudkowsky
- [Asterisk-Users] 2003-06-10 CVS: softphone connection failures
Moshe Yudkowsky
- [Asterisk-Users] VXML?
Moshe Yudkowsky
- [Asterisk-Users] Text to Speech - Someone needs to do this
Moshe Yudkowsky
- [Asterisk-Users] Text to Speech - Someone needs to do this
Moshe Yudkowsky
- [Asterisk-Users] Enhanced queue app
Anton Yurchenko
- [Asterisk-Users] Using switch =>
Anton Yurchenko
- [Asterisk-Users] Using switch =>
Anton Yurchenko
- [Asterisk-Users] Using switch =>
Anton Yurchenko
- [Asterisk-Users] Using switch =>
Anton Yurchenko
- [Asterisk-Users] Using switch => BUG?
Anton Yurchenko
- [Asterisk-Users] Using switch => SOLVED !!!!!! ( probably)
Anton Yurchenko
- [Asterisk-Users] ~3 seconds of silence when picking up a call
Anton Yurchenko
- [Asterisk-Users] "immediate=yes or Compleate recieved" with intcoming calls with new
CVS
Anton Yurchenko
- [Asterisk-Users] "immediate=yes or Compleate recieved" with intcoming
calls with new CVS
Anton Yurchenko
- [Asterisk-Users] Compleate recieved
Anton Yurchenko
- [Asterisk-Users] Back-to-back connected boards load test
Alex Zarubin
- [Asterisk-Users] TE410P startup (2 boards)
Alex Zarubin
- [Asterisk-Users] Mutex problem in sip?
Alex Zarubin
- [Asterisk-Users] Runtime error: Undefined symbol, have fetched new CVS and recompiled everything
Peter Zeltins
- [Asterisk-Users] SIP show channels display
Peter Zeltins
- [Asterisk-Users] AVM Fritz! to connect LAN with ISDN line?
Peter Zeltins
- [Asterisk-Users] AVM Fritz! to connect LAN with ISDN line?
Peter Zeltins
- [Asterisk-Users] AVM Fritz! to connect LAN with ISDN line?
Peter Zeltins
- [Asterisk-Users] chan_capi and poor voice quality
Peter Zeltins
- [Asterisk-Users] isdn4linux
Peter Zeltins
- [Asterisk-Users] ISDN Fritz & RedHat 8.0
Peter Zeltins
- [Asterisk-Users] Bugetone SIP Transfer
William Zhang
- [Asterisk-Users] New budgetone firmware
William Zhang
- [Asterisk-Users] three way calling and cisco ata 186
Pavel Zheltouhov
- [Asterisk-Users] three way calling and cisco ata 186
Pavel Zheltouhov
- [Asterisk-Users] three way calling and cisco ata 186
Pavel Zheltouhov
- [Asterisk-Users] mgcp problems
Pavel Zheltouhov
- [Asterisk-Users] mgcp problems
Pavel Zheltouhov
- [Asterisk-Users] mgcp problems
Pavel Zheltouhov
- [Asterisk-Users] mgcp problems
Pavel Zheltouhov
- [Asterisk-Users] asterisk,ata186 and Panasonic TD1232
Pavel Zheltouhov
- [Asterisk-Users] Problems with Hangup Detection in VoiceMail2.
Fred Ziegler
- [Asterisk-Users] how to set specific codec ?
Fernando Zuluaga
- [Asterisk-Users] Voicemail() problems - Long pause after incoming message
recording ended.
asterisk-users at sensecompute.com
- [Asterisk-Users] H.323 Gateway Connection
asterisk at ebctech.com
- [Asterisk-Users] How do i make best use of Macro?
asterisk at ebctech.com
- [Asterisk-Users] chan_h323, Asterisk and DTMF issue
asterisk at ebctech.com
- [Asterisk-Users] Help with ON-Hold, and call-transfer.
asterisk at taylorz.org.uk
- [Asterisk-Users] X100P in Australia
hafeez bana
- [Asterisk-Users] X100P in Australia
hafeez bana
- [Asterisk-Users] How do i make best use of Macro?
critch at basesys.com
- [Asterisk-Users] module : cdr_sybase.so
cvasiliu
- [Asterisk-Users] Asterisk SIP + Grandstream 100 phone
david at melita.net
- [Asterisk-Users] Call Transfer, Budgettone 100
david at melita.net
- [Asterisk-Users] (no subject)
deepak at mittals.com
- [Asterisk-Users] Migration to Asterisk - Running off of
Merlin Legend system
denon
- [Asterisk-Users] SIP immediate hangups with latest CVS
denon
- [Asterisk-Users] Phone System Questions
denon
- [Asterisk-Users] Call Transfer, Budgettone 100
denon
- [Asterisk-Users] Accounting info for SIP Calls
destan
- [Asterisk-Users] Dialogic
diana
- [Asterisk-Users] Cisco 7960g
dlah at tis.hr
- [Asterisk-Users] ENUM lookups
dlist
- [Asterisk-Users] AGI script sample using bash shell script
faizan at tilizone.com
- [Asterisk-Users] E1-RJ45 pin configuration
denzel fernando
- [Asterisk-Users] channel bank setup "What do I do now?"
firedude at shorelinuxsolutions.com
- [Asterisk-Users] channel bank configuration
firedude at shorelinuxsolutions.com
- [Asterisk-Users] channel bank configuration
firedude at shorelinuxsolutions.com
- [Asterisk-Users] slightly OT /how to obtain 900 number
firedude at shorelinuxsolutions.com
- [Asterisk-Users] outgoing callerid string
firedude at shorelinuxsolutions.com
- [Asterisk-Users] H3500CW recommendation
firedude at shorelinuxsolutions.com
- [Asterisk-Users] H3500CW recommendation
firedude at shorelinuxsolutions.com
- [Asterisk-Users] H3500CW recommendation
firedude at shorelinuxsolutions.com
- [Asterisk-Users] how do I do s extensions with PRI
firedude at shorelinuxsolutions.com
- [Asterisk-Users] how do I do s extensions with PRI
firedude at shorelinuxsolutions.com
- [Asterisk-Users] can't get musiconhold to work
firedude at shorelinuxsolutions.com
- [Asterisk-Users] can't get musiconhold to work
firedude at shorelinuxsolutions.com
- [Asterisk-Users] can't get musiconhold to work
firedude at shorelinuxsolutions.com
- [Asterisk-Users] can't get musiconhold to work
firedude at shorelinuxsolutions.com
- [Asterisk-Users] can't get musiconhold to work
firedude at shorelinuxsolutions.com
- [Asterisk-Users] Vonage
firedude at shorelinuxsolutions.com
- [Asterisk-Users] Is there any real asterisk documentation ?
frank.barthe at prescom.net
- [Asterisk-Users] PCI Master abort (debian / S400P & X101P
frank.barthe at prescom.net
- [Asterisk-Users] FXS PCI pb
frank.barthe at prescom.net
- [Asterisk-Users] PCI Master Abort
frank.barthe at prescom.net
- [Asterisk-Users] SUMMARY: Problems with Hangup Detection in VoiceMail2.
fred.ziegler at alum.mit.edu
- [Asterisk-Users] Echo on incoming calls (PRI->SIP) but not on outgoing (SIP->PRI)
fredrik.hedberg at telavox.se
- [Asterisk-Users] FW: Echo on incoming calls (PRI->SIP) but not on outgoing (SIP->PRI) (no html crap, sorry)
fredrik.hedberg at telavox.se
- [Asterisk-Users] Asterisk PBX Billing
shepherd fungayi
- [Asterisk-Users] Accurate Billing
shepherd fungayi
- [Asterisk-Users] Cellphone as an exchange line
shepherd fungayi
- [Asterisk-Users] Drop the call in 10min
isamar at isamarmaia.org
- [Asterisk-Users] GOTO inside AGI
isamar at isamarmaia.org
- [Asterisk-Users] Dial SessionTime
isamar at isamarmaia.org
- [Asterisk-Users] H323/No one is available to answer at this time
isamar at isamarmaia.org
- [Asterisk-Users] H323/No one is available to answer at this time
isamar at isamarmaia.org
- [Asterisk-Users] questions
isamar at isamarmaia.org
- [Asterisk-Users] questions
isamar at isamarmaia.org
- [Asterisk-Users] chan_h323
isamar at isamarmaia.org
- [Asterisk-Users] G729 doesn't work.
isamar at isamarmaia.org
- [Asterisk-Users] G729 doesn't work.
isamar at isamarmaia.org
- [Asterisk-Users] h323 and oh323 modules
isamar at isamarmaia.org
- [Asterisk-Users] the 'pound' and '#' are the same?
isamar at isamarmaia.org
- [Asterisk-Users] Microsoft SQL
isamar at isamarmaia.org
- [Asterisk-Users] Dialogic
isamar at isamarmaia.org
- [Asterisk-Users] MGCP behind NAT
ishpreet at optonline.net
- [Asterisk-Users] New Member
jltaylor
- [Asterisk-Users] Line Override Device
jltaylor
- [Asterisk-Users] EZ-Install
jltaylor
- [Asterisk-Users] EZ-Install
jltaylor
- [Asterisk-Users] EZ-Install
jltaylor
- [Asterisk-Users] Asterisk on Cygwin?
jltaylor
- [Asterisk-Users] Asterisk on Cygwin?
jltaylor
- [Asterisk-Users] OT: list format vs newsgroup format
jltaylor
- [Asterisk-Users] asterisk and uclinux
johncn
- [Asterisk-Users] Best E1 channel bank?
johncn
- [Asterisk-Users] h323 and oh323 modules
johncn
- [Asterisk-Users] the 'pound' and '#' are the same?
johncn
- [Asterisk-Users] Channel Language
johncn
- [Asterisk-Users] A solution for SIP and NAT
justin at vergeworks.com
- [Asterisk-Users] Asterisk crashes when trying to load G.729
module.
justin at vergeworks.com
- [Asterisk-Users] Ideal Prompt Recording Setup?
justin at vergeworks.com
- [Asterisk-Users] VoIP in hotels
robb kinnin
- [Asterisk-Users] isdn4linux/Teles16.3
l.heer at gmx.de
- [Asterisk-Users] Call Transfer Anouncement
listasterisk at comtek.co.nz
- [Asterisk-Users] Self parked but avaliable
listasterisk at comtek.co.nz
- [Asterisk-Users] Phone System Questions
lists
- [Asterisk-Users] Question About VOIP
lists
- [Asterisk-Users] Question #3
lists
- [Asterisk-Users] Question #3
lists
- [Asterisk-Users] Sip drops
lists
- [Asterisk-Users] Question on peer to peer config
lists
- [Asterisk-Users] How do I make Asterisk login at/use VoIP provider?
BK [address only for mailing lists]
- [Asterisk-Users] Asterisk Sacrifice?
BK [address only for mailing lists]
- [Asterisk-Users] Asterisk Sacrifice?
BK [address only for mailing lists]
- [Asterisk-Users] Asterisk Sacrifice?
BK [address only for mailing lists]
- [Asterisk-Users] CDR Information and Pipes
BK [address only for mailing lists]
- [Asterisk-Users] Please help -- Syntax for dialing VoIP provider
BK [address only for mailing lists]
- [Asterisk-Users] Please help -- Syntax for dialing VoIP provider
BK [address only for mailing lists]
- [Asterisk-Users] Please help -- Syntax for dialing VoIP provider
BK [address only for mailing lists]
- [Asterisk-Users] Follow-up -- Using Asterisk with Nikotel
BK [address only for mailing lists]
- [Asterisk-Users] Asterisk crashing after Voicemail box creation
BK [address only for mailing lists]
- [Asterisk-Users] Lot's of errors and warnings.
BK [address only for mailing lists]
- [Asterisk-Users] Using multiple iconnecthere accounts
BK [address only for mailing lists]
- [Asterisk-Users] ignorepat doesn't work
BK [address only for mailing lists]
- [Asterisk-Users] It's true - Nikotel charge for not-completed calls
BK [address only for mailing lists]
- [Asterisk-Users] ignorepat doesn't work
BK [address only for mailing lists]
- [Asterisk-Users] callerid= being ignored
BK [address only for mailing lists]
- [Asterisk-Users] incoming callerid on FXO
BK [address only for mailing lists]
- [Asterisk-Users] callerid= being ignored
BK [address only for mailing lists]
- [Asterisk-Users] It's true - Nikotel charge for not-completed calls
BK [address only for mailing lists]
- [Asterisk-Users] How to modify dialed number?
BK [address only for mailing lists]
- [Asterisk-Users] How to modify dialed number?
BK [address only for mailing lists]
- [Asterisk-Users] What does "callerid=" in sip.conf do?
BK [address only for mailing lists]
- [Asterisk-Users] Weird experience with MOH
BK [address only for mailing lists]
- [Asterisk-Users] Weird experience with MOH
BK [address only for mailing lists]
- [Asterisk-Users] What does "callerid=" in sip.conf do?
BK [address only for mailing lists]
- [Asterisk-Users] VIP 30 phone
BK [address only for mailing lists]
- [Asterisk-Users] Hardware / resources
marrandy
- [Asterisk-Users] Digital phones
marrandy
- [Asterisk-Users] Digital phones
marrandy
- [Asterisk-Users] Digital phones
marrandy
- [Asterisk-Users] Digital phones
marrandy
- [Asterisk-Users] Accurate Billing
marrandy
- [Asterisk-Users] Virtual fax on the Asterisk box
marrandy
- [Asterisk-Users] Lot's of errors and warnings.
marrandy
- [Asterisk-Users] Lot's of errors and warnings.
marrandy
- [Asterisk-Users] Lot's of errors and warnings.
marrandy
- [Asterisk-Users] Lot's of errors and warnings.
marrandy
- [Asterisk-Users] Lot's of errors and warnings.
marrandy
- [Asterisk-Users] Lot's of errors and warnings.
marrandy
- [Asterisk-Users] voip
marrandy
- [Asterisk-Users] voip
marrandy
- [Asterisk-Users] Asterisk basic how-to on O'Reilly's site
marrandy
- [Asterisk-Users] Transfers on the Cisco 7960
marrandy
- [Asterisk-Users] Why mp3 (licensing issues) as opposed to Open Source OGG
marrandy
- [Asterisk-Users] Why mp3 (licensing issues) as opposed to Open Source OGG
marrandy
- [Asterisk-Users] grandstream sip phone
marrandy
- [Asterisk-Users] X100P - FO card
marrandy
- [Asterisk-Users] Offering an Asterisk Documentation and FAQ Portal
marrandy
- [Asterisk-Users] Grandstream Budgettone 100 & 102
marrandy
- [Asterisk-Users] T1 config for robbed-bit E&M AMI
mattf
- [Asterisk-Users] Cisco 7910 compatibility
mattf
- [Asterisk-Users] Techfone VOIP phone
mattf
- [Asterisk-Users] Dial plan doesn't seem to save properly
mvickers at real.com
- [Asterisk-Users] insmod wcfxo failed ( b8zs, esf, wink start is what I'm trying to
do.)
mvickers at real.com
- [Asterisk-Users] Re: insmod wcfxo failed ( b8zs, esf, wink start is what I'm trying
to do.)
mvickers at real.com
- [Asterisk-Users] Music on hold quality..
nathan
- [Asterisk-Users] Vonage
nathan
- [Asterisk-Users] Seg Fault!!
rafa
- [Asterisk-Users] overlap dialing on a pri span
salmon at netzquadrat.de
- [Asterisk-Users] Busy detect on pri channel?
salmon at netzquadrat.de
- [Asterisk-Users] Asterisk - Protocol Converter from SIP/H.323
sam
- [Asterisk-Users] capi_chan error - CAPI not loaded.
Peer Oliver schmidt
- [Asterisk-Users] capi_chan error - CAPI not loaded.
Peer Oliver schmidt
- [Asterisk-Users] chan_capi and poor voice quality
Peer Oliver schmidt
- [Asterisk-Users] chan_capi and poor voice quality
Peer Oliver schmidt
- [Asterisk-Users] Debian Package asterisk-oh323?
Peer Oliver schmidt
- [Asterisk-Users] SIP/H.323 Phone with intercom
Peer Oliver schmidt
- [Asterisk-Users] AVM C4 and external and internal ISDN bus and *
Peer Oliver schmidt
- [Asterisk-Users] Configuration sample for isdn4linux?
Peer Oliver schmidt
- [Asterisk-Users] * behind ISDN pbx - Forwarding to extensions with in primary pbx
Peer Oliver schmidt
- [Asterisk-Users] Channel Language
Peer Oliver schmidt
- [Asterisk-Users] Channel Language
Peer Oliver schmidt
- AW: [Asterisk-Users] * behind ISDN pbx - Forwarding to extensions
with in primary pbx
Peer Oliver schmidt
- [Asterisk-Users] Channel Language
Peer Oliver schmidt
- AW: [Asterisk-Users] * behind ISDN pbx - Forwarding to extensions
with in primary pbx
Peer Oliver schmidt
- [Asterisk-Users] Channel Language
Peer Oliver schmidt
- [Asterisk-Users] The Budgetone 100
shido
- [Asterisk-Users] AUSTEL Certified
shido
- [Asterisk-Users] Question #3
shido
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality
speech synthesis?
sjacobs at rugburn.org
- [Asterisk-Users] Poll - Would you pay $30-$50 for high quality
speech synthesis?
sjacobs at rugburn.org
- [Asterisk-Users] Problems getting 7960's to play nice with Asterisk
sjacobs at rugburn.org
- [Asterisk-Users] Problems getting 7960's to play nice with
Asterisk
sjacobs at rugburn.org
- [Asterisk-Users] Problems getting 7960's to play nice with
Asterisk
sjacobs at rugburn.org
- [Asterisk-Users] Problems getting 7960's to play nice with
Asterisk
sjacobs at rugburn.org
- [Asterisk-Users] how to start
sohail at mysuperphone.com
- [Asterisk-Users] Activate MySQL logging
surajee at infotechs.lk
- [Asterisk-Users] Activate MySQL logging
surajee at infotechs.lk
- [Asterisk-Users] Activate MySQL logging
surajee at infotechs.lk
- [Asterisk-Users] Accurate Billing
surajee at infotechs.lk
- [Asterisk-Users] Accurate Billing
surajee at infotechs.lk
- [Asterisk-Users] ISDN PRI E1 configuration with E100P
surajee at infotechs.lk
- [Asterisk-Users] ISDN PRI E1 configuration with E100P
surajee at infotechs.lk
- [Asterisk-Users] ISDN PRI E1 configuration with E100P
surajee at infotechs.lk
- [Asterisk-Users] ISDN PRI E1 configuration with E100P
surajee at infotechs.lk
- [Asterisk-Users] ISDN PRI E1 configuration with E100P
surajee at infotechs.lk
- [Asterisk-Users] ISDN PRI E1 configuration with E100P
surajee at infotechs.lk
- [Asterisk-Users] Timeout in Call Transfering
surajee at infotechs.lk
- [Asterisk-Users] Call Transfer
surajee at infotechs.lk
- [Asterisk-Users] Enhanced queue app
tim.mcqueen at qualisys.biz
- [Asterisk-Users] Hardware Vendors
tmassey at obscorp.com
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
tmassey at obscorp.com
- [Asterisk-Users] "Best" VoIP provider for Asterisk?
tmassey at obscorp.com
- [Asterisk-Users] questions
tmassey at obscorp.com
- [Asterisk-Users] Summary of VoIP options for Asterisk and request for more?
tmassey at obscorp.com
- [Asterisk-Users] Vonage
tmassey at obscorp.com
- [Asterisk-Users] extensions
tns
- [Asterisk-Users] client reinvitation problem
vk at akcecc.net
- [Asterisk-Users] client reinvitation problem
vk at akcecc.net
- [Asterisk-Users] ATA-186 re INVITE message turn off
vk at akcecc.net
- [Asterisk-Users] voip
wasim at convergence.com.pk
- [Asterisk-Users] Best E1 channel bank?
wasim at convergence.com.pk
- [Asterisk-Users] can't get musiconhold to work
wipeout at linuxmail.org
- [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
yves.schaaf at restena.lu
- [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
yves.schaaf at restena.lu
Last message date:
Thu Jul 31 22:28:56 MST 2003
Archived on: Tue Sep 5 15:25:40 MST 2006
This archive was generated by
Pipermail 0.09 (Mailman edition).