Asterisk Internet Phone System

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Asterisk 1.8 vici and the fax, SMS, gtalk, Jaber channels, bilal ghayyad
Performance Asterisk large installation on Vmware/Xen, Rafael dos Santos Saraiva
- Re: Performance Asterisk large installation on Vmware/Xen, Mitul Limbani
  - Re: Performance Asterisk large installation on Vmware/Xen, Chris Bagnall
    - Re: Performance Asterisk large installation on Vmware/Xen, Mike
- Re: Performance Asterisk large installation on Vmware/Xen, James Cloos
- Re: Performance Asterisk large installation on Vmware/Xen, Stelios Koroneos
Asterisk 1.8-cert and AGC, Maximilian Grobecker
- Re: Asterisk 1.8-cert and AGC, Matthew Jordan
Asterisk 11.4.0 Now Available, Asterisk Development Team
Asterisk 1.8.22.0 Now Available, Asterisk Development Team
Auto dialer scripts and software, cjwstudios
- Re: Auto dialer scripts and software, Carlos Chavez
Temporarily features (transfer) off during Read, Dmitriy Serov
wanpipe and digium, oslec and hardware echo canceller, bilal ghayyad
- Re: wanpipe and digium, oslec and hardware echo canceller, Angelo Delphini
  - Re: wanpipe and digium, oslec and hardware echo canceller, jg
Planned maintenance for community services on May 16, 2013, Asterisk Development Team
Re: asterisk-users Digest, Vol 106, Issue 23, Nicholas Hart
Asterisk High-availability/failover solutions, Andre Goree
11.4: motif can only handle one channel at a time?, sean darcy
- Re: 11.4: motif can only handle one channel at a time?, Joshua Colp
- Re: 11.4: motif can only handle one channel at a time?, sean darcy
AstriCon 2013 (our 10th AstriCon) needs YOU!, David Duffett
Call Transfer question, Muhammad Faheem
- Re: Call Transfer question, qasimakhan@xxxxxxxxx
Re: Asterisk Web Meetme module not loading, Rohit Mahajan
- Re: Asterisk Web Meetme module not loading, Dan Austin
SetCallerPres questions, Adam Moffett
- Re: SetCallerPres questions, Maximilian Grobecker
Polycom and forwarding., Ken D'Ambrosio
- Re: Polycom and forwarding., Richard Mudgett
- Re: Polycom and forwarding., Carlos Alvarez
Cut offs on outgoing SIP calls, Daniel - Asterisk
- Re: Cut offs on outgoing SIP calls, Gertjan Baarda
  - Re: Cut offs on outgoing SIP calls, Daniel - Asterisk
- Re: Cut offs on outgoing SIP calls, Asghar Mohammad
  - Re: Cut offs on outgoing SIP calls, Daniel - Asterisk
    - Re: Cut offs on outgoing SIP calls, Asghar Mohammad
      - Re: Cut offs on outgoing SIP calls, Daniel - Asterisk
        
        Re: Cut offs on outgoing SIP calls, Asghar Mohammad
- <Possible follow-ups>
- Re: Cut offs on outgoing SIP calls, Philipp von Klitzing
How to allow AMI access to Originate yet deny Application: System, Alex Villacís Lasso
Re: 3. mfcr2 channel state IDLE 0x00 and call trace log file not ended ?? (Leonardo Rivanera), Mc GRATH Ricardo
Using PHPMyAdmin to remotely access Asterisk MySQL Database, Lobna Hegazy
- Re: Using PHPMyAdmin to remotely access Asterisk MySQL Database, Gertjan Baarda
  - Re: Using PHPMyAdmin to remotely access Asterisk MySQL Database, Lobna Hegazy
    - Re: Using PHPMyAdmin to remotely access Asterisk MySQL Database, Gertjan Baarda
      - Re: Using PHPMyAdmin to remotely access Asterisk MySQL Database, Lobna Hegazy
        
        Re: Using PHPMyAdmin to remotely access Asterisk MySQL Database, Brian LaVallee
      - Initial REGISTER Request: Contains Credentials before 401, Brian LaVallee
        
        Re: Initial REGISTER Request: Contains Credentials before 401, Matthew J. Roth
        
        Initial REGISTER Request: Contains Credentials before 401: KDDI Japan, Brian LaVallee
        
        Re: Initial REGISTER Request: Contains Credentials before 401: KDDI Japan, Matthew J. Roth
- Re: Using PHPMyAdmin to remotely access Asterisk MySQL Database, Asghar Mohammad
mfcr2 channel state IDLE 0x00 and call trace log file not ended ??, Leonardo Rivanera
dial and bridge, Lenz Emilitri
- Re: dial and bridge, Mitul Limbani
  - Re: dial and bridge, Lenz Emilitri
    - Re: dial and bridge, Mitul Limbani
      - Re: dial and bridge, Lenz Emilitri
        
        Re: dial and bridge, Ioan Indreias
        
        Re: dial and bridge, Ioan Indreias
- Re: dial and bridge, Warren Selby
  - Re: dial and bridge, Lenz Emilitri
- Re: dial and bridge, Mitul Limbani
- Re: dial and bridge, Dan Cropp
  - Re: dial and bridge, Lenz Emilitri
Monitoring SIP trunk status on call by call basis, Ishfaq Malik
- Re: Monitoring SIP trunk status on call by call basis, Asghar Mohammad
- Re: Monitoring SIP trunk status on call by call basis, Chris Bagnall
Call Diversion Override, Dominik George
Asterisk 11.3 and Kamailio 4.0 Realtime Integration Tutorial, Daniel-Constantin Mierla
- Re: Asterisk 11.3 and Kamailio 4.0 Realtime Integration Tutorial, Matthew Jordan
Upgrade from 1.0.x to AsteriskNOW 3.0, Andre Goree
- Re: Upgrade from 1.0.x to AsteriskNOW 3.0, Dennis Dryden
  - Re: Upgrade from 1.0.x to AsteriskNOW 3.0, Andre Goree
    - Re: Upgrade from 1.0.x to AsteriskNOW 3.0, Kevin Larsen
amiDebugger - might make your life easier if you program through the AMI, Lenz Emilitri
- Re: amiDebugger - might make your life easier if you program through the AMI, Yves A.
Sangoma Wanpipe Driver, Yves A.
- Re: Sangoma Wanpipe Driver, Duncan Turnbull
  - Re: Sangoma Wanpipe Driver, Yves A.
    - Re: Sangoma Wanpipe Driver, Asghar Mohammad
      - Re: Sangoma Wanpipe Driver, Yves A.
        
        Re: Sangoma Wanpipe Driver, Yves A.
Integrate Astreisk with SIP interface, luke devon
- Re: Integrate Astreisk with SIP interface, longst
  - Re: Integrate Astreisk with SIP interface, luke devon
- Re: Integrate Astreisk with SIP interface, Asghar Mohammad
  - Re: Integrate Astreisk with SIP interface, Steve Edwards
time zone setting in asterisk, Joseph
- Re: time zone setting in asterisk, Asghar Mohammad
  - Re: time zone setting in asterisk, Joseph
    - Re: time zone setting in asterisk, Asghar Mohammad
      - Re: [SOLVED] time zone setting in asterisk, Joseph
- <Possible follow-ups>
- Re: time zone setting in asterisk, Gregory Malsack
  - Re: time zone setting in asterisk, Joseph
- Re: time zone setting in asterisk, Gregory Malsack
HD Voice -- connecting Asterisk into HD Voice compatible mobile phone, Brandon B.
Which channels are required for FAX, GTALK and Jaber, bilal ghayyad
- AT&T uverse Motorolga nvg510, David Wessell
AMI Originate issue, Muhammad Faheem
- Re: AMI Originate issue, Matthew Jordan
dahdi driver not getting install, Harish Mandowara
- Re: dahdi driver not getting install, Asghar Mohammad
- Re: dahdi driver not getting install, Alec Davis
  - Re: dahdi driver not getting install, Asghar Mohammad
  - Re: dahdi driver not getting install, Andrew Colin
    - Re: dahdi driver not getting install, Asghar Mohammad
      - Re: dahdi driver not getting install, Andrew Colin
        
        Re: dahdi driver not getting install, Salaheddine Elharit
        
        Re: dahdi driver not getting install, Patrick Lists
11.4: no incoming gv/xmpp, sean darcy
Tier 1 Service Providers (AT&T, Level 3), Nick Khamis
- Re: Tier 1 Service Providers (AT&T, Level 3), Nick Khamis
  - Message not available
    - Re: Tier 1 Service Providers (AT&T, Level 3), Nick Khamis
ISP trunk session ID?, Sergej Petrovsky
- Re: ISP trunk session ID?, Asghar Mohammad
  - Re: ISP trunk session ID?, Nick Khamis
    - Re: ISP trunk session ID?, Asghar Mohammad
      - Re: ISP trunk session ID?, Sergej Petrovsky
        
        Re: ISP trunk session ID [SOLVED], Sergej Petrovsky
Asterisk 12 and OPUS Codec, James Mortensen
- Re: Asterisk 12 and OPUS Codec, Paul Belanger
qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`, Brian LaVallee
- Re: qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`, Jeremy Kister
  - Re: qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`, Nathan Anderson
  - Thanks! qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`, Brian LaVallee
DID providers, Jeff LaCoursiere
Planned maintenance for community services on May 11, 2013, Asterisk Development Team
monitoring Asterisk 1.8, motty cruz
- Re: monitoring Asterisk 1.8, Carlos Alvarez
  - Re: monitoring Asterisk 1.8, motty cruz
    - Re: monitoring Asterisk 1.8, Carlos Alvarez
- Re: monitoring Asterisk 1.8, Carlos Rojas
  - Re: monitoring Asterisk 1.8, Carlos Rojas
    - Re: monitoring Asterisk 1.8, motty cruz
      - Re: monitoring Asterisk 1.8, Carlos Alvarez
      - Re: monitoring Asterisk 1.8, Carlos Rojas
      - Re: monitoring Asterisk 1.8, Bruce Reeves
- Re: monitoring Asterisk 1.8, Jai Rangi
- Re: monitoring Asterisk 1.8, Michel Verbraak
  - Re: monitoring Asterisk 1.8, Luis Morales
    - Voicemail send to e-mail, Bory's Rouliane Kouassi
chanstats console errors, asterisk-02
- Re: chanstats console errors, Jeremy Kister
question about CDR, Salaheddine Elharit
- Re: question about CDR, Ishfaq Malik
  - Re: question about CDR, Salaheddine Elharit
- Re: question about CDR, Asghar Mohammad
  - Re: question about CDR, Salaheddine Elharit
No early media on 302 redirect via two servers, Carlos Alvarez
Transfer cmd via AsyncAGI, Dan Cropp
Confbridge Dynamic video_mode, Rizwan Hisham
Obtaining high voice quality in dahdi channel, bilal ghayyad
- Re: Obtaining high voice quality in dahdi channel, jg
Asterisk and hylafax: how to debug ..., Sebastian Niehaus
- Re: Asterisk and hylafax: how to debug ..., Sebastian Niehaus
  - Re: Asterisk and hylafax: how to debug ..., Karsten Wemheuer
- Re: Asterisk and hylafax: how to debug ..., James Cloos
passing '302 moved temporarily' back to the SIP provider, Johann Steinwendtner
- Re: passing '302 moved temporarily' back to the SIP provider, Satish Barot
  - Re: passing '302 moved temporarily' back to the SIP provider, Carlos Alvarez
    - Re: passing '302 moved temporarily' back to the SIP provider, Satish Barot
      - Re: passing '302 moved temporarily' back to the SIP provider, Satish Barot
Get Channel Variables in AMI Event NewExten, Faheem
- <Possible follow-ups>
- Re: Get Channel Variables in AMI Event NewExten, Dan Cropp
  - Re: Get Channel Variables in AMI Event NewExten, Matthew Jordan
    - Re: Get Channel Variables in AMI Event NewExten, Muhammad Faheem
НА: asterisk-users Digest, Vol 105, Issue 40, virus.chel@mail.ru
- Re: НА: asterisk-users Digest, Vol 105, Issue 40, Hans Witvliet
chan_alsa and confbridge, Chris Gentle
- Re: chan_alsa and confbridge, Chris Gentle
What is bootstrap.sh for ? Possible bug in 11.3.0 ?, Olivier
- Re: What is bootstrap.sh for ? Possible bug in 11.3.0 ?, Matthew Jordan
  - Re: What is bootstrap.sh for ? Possible bug in 11.3.0 ?, Olivier
    - Re: What is bootstrap.sh for ? Possible bug in 11.3.0 ?, Jason Parker
      - Re: What is bootstrap.sh for ? Possible bug in 11.3.0 ?, Olivier
Installing on an OpenVZ instance, James Wystead
- Re: Installing on an OpenVZ instance, Johan Wilfer
MRCPSynth() change voice, Grant Bagdasarian
OT - Question on Aastra 6735i - Was: Sip phone displaying caller name while on call, Olivier
OT - Differences between Aastra 6730i and 6750i series, Olivier
- Re: OT - Differences between Aastra 6730i and 6750i series, Rusty Newton
Joining an astablished call, neo haux
- Re: Joining an astablished call, Ian Pilcher
- Re: Joining an astablished call, Alec Davis
- Re: Joining an astablished call, John Novack
  - Re: Joining an astablished call, Jacob . E . Miles
Testing 911 call, Joseph
- Re: Testing 911 call, Mark Engelhardt
  - Re: Testing 911 call, Dale Noll
    - Re: Testing 911 call, James Miller
      - Re: Testing 911 call, David Wessell
Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'", Sandeep Raju
- Re: Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'", Alec Davis
  - Re: Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'", Sandeep Raju
    - Re: Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'", Sandeep Raju
      - Re: Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'", Alec Davis
GotoIf DIALSTATUS - not working, Joseph
- Re: GotoIf DIALSTATUS - not working, Alec Davis
  - Re: GotoIf DIALSTATUS - not working, Joseph
Re: BLF and asterisk Queue, Alec Davis
11.4.-rc1: new segfault in iksemel ??, sean darcy
My new Polycom 450's can't xfer to 4-digit extension, Mike Diehl
- Re: My new Polycom 450's can't xfer to 4-digit extension, Doug Lytle
- Re: My new Polycom 450's can't xfer to 4-digit extension, Dave Fullerton
AMI help needed, Pat Collins
- Re: AMI help needed, Faheem
Cisco 9971 help, Patrick Lidstone
- Re: Cisco 9971 help, Stoyan Marinov
- <Possible follow-ups>
- Re: Cisco 9971 help, Patrick Lidstone
  - Re: Cisco 9971 help, Patrick Lidstone
    - Re: Cisco 9971 help, Patrick Lidstone
Digium D70 visual voicemail - won't play, Dr. Michael J. Chudobiak
changing ringtones to a group of phones, Dr. Michael J. Chudobiak
- Re: changing ringtones to a group of phones, jg
  - Re: changing ringtones to a group of phones, Dr. Michael J. Chudobiak
VoIP Incoming Issue, Gopalakrishnan N
Playing a sound file during a call, Carlos Alvarez
- Re: Playing a sound file during a call, Kevin Larsen
  - Re: Playing a sound file during a call, Carlos Alvarez
    - Re: Playing a sound file during a call, Kevin Larsen
      - Re: Playing a sound file during a call, Carlos Alvarez
        
        Re: Playing a sound file during a call, Carlos Alvarez
        
        Re: Playing a sound file during a call, Richard Mudgett
        
        Re: Playing a sound file during a call, Michael L. Young
        
        Re: Playing a sound file during a call, Carlos Alvarez
Building Asterisk 11.4.0-rc1 with PJSIP 2.1, James Mortensen
- Re: Building Asterisk 11.4.0-rc1 with PJSIP 2.1, Yves A.
debug strategy for one-way audio calls, Marie Fischer
- Re: debug strategy for one-way audio calls, Johan Wilfer
  - Re: debug strategy for one-way audio calls, Gopalakrishnan N
- Re: debug strategy for one-way audio calls, Olivier
  - Re: debug strategy for one-way audio calls, Marie Fischer
    - Re: debug strategy for one-way audio calls, Olivier
Queues with different technologies for members, like Khomp Driver, Daniel Varella
Call "stuck" in queue, Mitch Claborn
- Re: Call "stuck" in queue, Mitch Claborn
SMS Scenario, bilal ghayyad
- Re: SMS Scenario, A J Stiles
Re: asterisk-users Digest, Vol 105, Issue 39, bipin singh
Re: multiple provider for incoming, Gregory Malsack
- Re: multiple provider for incoming, Matt Hamilton
  - Re: multiple provider for incoming, Eric Wieling
asterisk 1.4 and SMS module, bilal ghayyad
- multiple provider for incoming, Matt Hamilton
  - Re: multiple provider for incoming, David Wessell
    - Re: multiple provider for incoming, Warren Selby
      - Re: multiple provider for incoming, David Wessell
      - Re: multiple provider for incoming, Matt Hamilton
        
        Re: multiple provider for incoming, Don Kelly
        
        Re: multiple provider for incoming, Matt Hamilton
        
        Re: multiple provider for incoming, Don Kelly
        
        Re: multiple provider for incoming, Carlos Alvarez
Asterisk QSIG doesnt send the calling name to Nortel CS1000, Danilo Dionisi
- Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000, Dale Noll
  - Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000, Danilo Dionisi
    - Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000, Danilo Dionisi
      - Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000, Dale Noll
      - Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000, Johann Steinwendtner
- Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000, Richard Mudgett
  - Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000, Danilo Dionisi
    - Re: Asterisk QSIG doesnt send the calling name to Nortel CS1000, Dale Noll
Gateway?, James Wystead
- Re: Gateway?, jg
- Re: Gateway?, A J Stiles
  - Re: Gateway?, Eric Wieling
    - Re: Gateway?, James Wystead
      - Re: Gateway?, Don Kelly
        
        Re: Gateway?, Asghar Mohammad
      - Re: Gateway?, jg
Asterisk 11.3.0 - Mask for new file not correct, Ludovic Boué
- Re: Asterisk 11.3.0 - Mask for new file not correct, David M. Lee
  - Re: Asterisk 11.3.0 - Mask for new file not correct, Ludovic Boué
Can't register to Asterisk 1.6 with old Aastra phones, Carlos Alvarez
- Re: Can't register to Asterisk 1.6 with old Aastra phones, Nathan Anderson
- Re: Can't register to Asterisk 1.6 with old Aastra phones, Patrick Lists
  - Re: Can't register to Asterisk 1.6 with old Aastra phones, Carlos Alvarez
- Re: Can't register to Asterisk 1.6 with old Aastra phones, Bob Kyeyune
glibc detected crash, Kelly Opal
caller_id vs cid_number, Nick Khamis
looking for a way to do appointment reminders, Brandon Coale
- Re: looking for a way to do appointment reminders, Yves A.
- Re: looking for a way to do appointment reminders, jg
  - Re: looking for a way to do appointment reminders, Hans Witvliet
    - Re: looking for a way to do appointment reminders, jg
      - Re: looking for a way to do appointment reminders, Chris Bagnall
        
        Re: looking for a way to do appointment reminders, Ron Wheeler
        
        Re: looking for a way to do appointment reminders, David stahl
        
        Re: looking for a way to do appointment reminders, Lenz Emilitri
        
        Re: looking for a way to do appointment reminders, Brandon Coale
    - Re: looking for a way to do appointment reminders, Chris Bagnall
      - Re: looking for a way to do appointment reminders, jg
        
        Re: looking for a way to do appointment reminders, Chris Bagnall
      - Re: looking for a way to do appointment reminders, jg
Users appending # sign when dialing an extension from automated greeting, Vernon Polinkichov
Load Balancing, Olivier
- Re: Load Balancing, acheraime
  - Re: Load Balancing, Olivier
Asterisk Calendar integration suggestions, john@xxxxxxxxxxx
- Re: Asterisk Calendar integration suggestions, Steve Totaro
  - Re: Asterisk Calendar integration suggestions, Hans Witvliet
Asterisk 11.4.0-rc1 refuses to use the TURN server, James Mortensen
Dialplan reload not reloading everything, Brandon Mackie
- Re: Dialplan reload not reloading everything, Rusty Newton
Jitter Buffer in asterisk 1.8.11.0, Muhammad Yousuf
- Re: Jitter Buffer in asterisk 1.8.11.0, qasimakhan@xxxxxxxxx
/dev/dahdi/pseudo leaking, Valter Nogueira
Asterisk Tech Job Posting Dallas Texas, JR Richardson
Planned maintenance for community services on April 22, 2013, Asterisk Development Team
H.264 high profile support, Jeff Morriss
Device states, Tommy Cooper
Installing Asterisk on Virtual Machine, Sandeep Raju
- Re: Installing Asterisk on Virtual Machine, Doug Lytle
  - Re: Installing Asterisk on Virtual Machine, Sandeep Raju
    - Re: Installing Asterisk on Virtual Machine, Doug Lytle
- Re: Installing Asterisk on Virtual Machine, Tzafrir Cohen
  - Re: Installing Asterisk on Virtual Machine, Sandeep Raju
    - Re: Installing Asterisk on Virtual Machine, Hans Witvliet
      - Re: Installing Asterisk on Virtual Machine, Sandeep Raju
        
        Re: Installing Asterisk on Virtual Machine, Sandeep Raju
        
        Re: Installing Asterisk on Virtual Machine, Sandeep Raju
        
        Re: Installing Asterisk on Virtual Machine, Tzafrir Cohen
        
        Re: Installing Asterisk on Virtual Machine, Sandeep Raju
        
        Re: Installing Asterisk on Virtual Machine, Sandeep Raju
        
        Re: Installing Asterisk on Virtual Machine, mailinglist
h323-sip: one way connection, s m
- Re: h323-sip: one way connection, Asghar Mohammad
  - Re: h323-sip: one way connection, s m
    - Re: h323-sip: one way connection, Asghar Mohammad
      - Re: h323-sip: one way connection, s m
        
        Re: h323-sip: one way connection, Asghar Mohammad
        
        Re: h323-sip: one way connection, s m
        
        Re: h323-sip: one way connection, Asghar Mohammad
        
        Re: h323-sip: one way connection, s m
        
        Re: h323-sip: one way connection, Asghar Mohammad
Strange problem with Asterisk 1.8.9.3, Dereck D
- Re: Strange problem with Asterisk 1.8.9.3, dotnetdub
  - Re: Strange problem with Asterisk 1.8.9.3, Dereck D
    - Re: Strange problem with Asterisk 1.8.9.3, Dereck D
    - Re: Strange problem with Asterisk 1.8.9.3, dotnetdub
CDR Question, Nyamul Hassan
- Re: CDR Question, jg
  - Re: CDR Question, Nyamul Hassan
    - Re: CDR Question, jg
      - Re: CDR Question, Nyamul Hassan
Dynamic realtime + queues.conf Unresolved, Tommy Cooper
- Re: Dynamic realtime + queues.conf Unresolved, Nathan Anderson
set google voice callerid as Unknown/Unavailable ?, sean darcy
Sip phone displaying caller name while on call, Olivier
E911 Voip Trunking, Chris Nighswonger
- Re: E911 Voip Trunking, Terry Brummell
  - Re: E911 Voip Trunking, Chris Nighswonger
- Re: E911 Voip Trunking, Warren Selby
  - Re: E911 Voip Trunking, Nathan Anderson
    - Re: E911 Voip Trunking, Chris Nighswonger
To enhance the voice quality of the SIP trunk, bilal ghayyad
Dynamic realtime + queues, Tommy Cooper
- <Possible follow-ups>
- Dynamic realtime + queues, Tommy Cooper
  - Re: Dynamic realtime + queues, Leandro Dardini
- Dynamic realtime + queues, Tommy Cooper
  - Re: Dynamic realtime + queues, Leandro Dardini
    - Message not available
      - Re: Dynamic realtime + queues, Tommy Cooper
        
        Re: Dynamic realtime + queues, Jose Flores Galicia
        
        Message not available
        
        Fw: Dynamic realtime + queues, Tommy Cooper
ODBC dialplan looping problem, Pat Collins
- Re: ODBC dialplan looping problem, Doug Lytle
  - Re: ODBC dialplan looping problem, Pat Collins
- Re: ODBC dialplan looping problem, Bharat Lalcheta
  - Re: ODBC dialplan looping problem, Pat Collins
    - Re: ODBC dialplan looping problem, Bharat Lalcheta
      - Re: ODBC dialplan looping problem, Pat Collins
- Re: ODBC dialplan looping problem, jg
  - Re: ODBC dialplan looping problem, Dale Noll
  - Re: ODBC dialplan looping problem, Matthew Jordan
- Re: ODBC dialplan looping problem, Satish Barot
  - Re: ODBC dialplan looping problem, Satish Barot
  - Sip and the media path, David Wessell
    - Re: Sip and the media path, Kevin Larsen
      - Re: Sip and the media path, David Wessell
        
        Re: Sip and the media path, Kevin Larsen
        
        Re: Sip and the media path, qasimakhan@xxxxxxxxx
How to show caller number ?, neo haux
- Re: How to show caller number ?, Steve Edwards
Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger, bilal ghayyad
- Re: Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger, isrlgb
  - Re: Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger, Lenz Emilitri
    - 回覆︰ Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger, kingman chui
      - Re: 回覆︰ Asterisk with whatsapp, facebook, viber, yahoo and hotmail messanger, Lenz Emilitri
Users.conf vs Sip.conf, Bryan Anderson
core console debug on single file, Gabriel Ortiz Lour
- Re: core console debug on single file, Richard Mudgett
Caller ID is not persisted when using Channel Redirect, Jacob . E . Miles
failed to extend from 512 to 676 on cli, Kamlesh Kumar
Phpagi action based on outbound call user response, Rahul R
- Re: Phpagi action based on outbound call user response, Lenz Emilitri
Transfer only, no outbound calling, Todd Routhier
- Re: Transfer only, no outbound calling, Nathan Anderson
  - Re: Transfer only, no outbound calling, Todd Routhier
On SIP INVITE answering to IP:port found in Contact: header., Markus
- Re: On SIP INVITE answering to IP:port found in Contact: header., Matthew J. Roth
- Re: On SIP INVITE answering to IP:port found in Contact: header., Joshua Colp
  - Re: On SIP INVITE answering to IP:port found in Contact: header., Matthew J. Roth
    - Re: On SIP INVITE answering to IP:port found in Contact: header., Markus
      - Re: On SIP INVITE answering to IP:port found in Contact: header., Matthew J. Roth
    - Re: On SIP INVITE answering to IP:port found in Contact: header., Joshua Colp
      - Re: On SIP INVITE answering to IP:port found in Contact: header., Matthew J. Roth
erro compiling dahdi, Jonas Kellens
- Re: erro compiling dahdi, Shaun Ruffell
- Re: erro compiling dahdi, Russ Meyerriecks
Access postgresql directly from dialplan?, Sebastian Arcus
- Re: Access postgresql directly from dialplan?, Gertjan Baarda
  - Re: Access postgresql directly from dialplan?, Sebastian Arcus
Traffic Crossover, Geoffrey Yeoh
- Re: Traffic Crossover, Eric Wieling
Asterisk SIP TCP, Zohair Raza
- Re: Asterisk SIP TCP, Mehroz Ashraf
  - Re: Asterisk SIP TCP, Mark Henry
    - Re: Asterisk SIP TCP, Mark Henry
      - Re: Asterisk SIP TCP, Zohair Raza
        
        Re: Asterisk SIP TCP, Bharat Lalcheta
        
        Re: Asterisk SIP TCP, Zohair Raza
        
        Re: Asterisk SIP TCP, Bharat Lalcheta
        
        Re: Asterisk SIP TCP, Zohair Raza
Dial multiple device cancellation, Santi Anton
- Re: Dial multiple device cancellation, jg
- Re: Dial multiple device cancellation, Satish Barot
- Re: Dial multiple device cancellation, Olivier
  - Re: Dial multiple device cancellation, Santi Anton
Polycom Soundpoint IP 330 provisioning, Daniel - Asterisk
- Re: Polycom Soundpoint IP 330 provisioning, Kevin Larsen
  - Re: Polycom Soundpoint IP 330 provisioning, Daniel - Asterisk
    - Re: Polycom Soundpoint IP 330 provisioning, Daniel - Asterisk
    - Re: Polycom Soundpoint IP 330 provisioning, Dave Fullerton
      - Re: Polycom Soundpoint IP 330 provisioning, Daniel - Asterisk
Network based transcoding, Nick Khamis
- Re: Network based transcoding, Eric Wieling
  - Re: Network based transcoding, jg
    - Re: Network based transcoding, Nick Khamis
      - Re: Network based transcoding, Chris Bagnall
      - Re: Network based transcoding, jg
        
        Re: Network based transcoding, Nick Khamis
        
        Re: Network based transcoding, Nick Khamis
Progress() on outgoing calls, Jonas Kellens
- Re: Progress() on outgoing calls, Rusty Newton
Voicemail Prepend not working properly on 1.8.18, James Lamanna
"Dropping call because extensions '200', 's' and 'i' doesn't exists", s m
- Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists", A J Stiles
  - Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists", s m
    - Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists", Asghar Mohammad
      - Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists", s m
        
        Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists", Asghar Mohammad
    - Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists", Gertjan Baarda
      - Re: "Dropping call because extensions '200', 's' and 'i' doesn't exists", s m
Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes, Thorsten Göllner
- Re: Asterisk 11.2.1 / dahdi destroy channel / asteriskcrashes, Alec Davis
  - Re: Asterisk 11.2.1 / dahdi destroy channel / asteriskcrashes, Richard Mudgett
- PRI DEBUG, Yves A.
  - Re: PRI DEBUG, Richard Mudgett
    - Re: PRI DEBUG, Yves A.
- Re: Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes, Yves A.
  - Re: Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes, Thorsten Göllner
Setting a CDR field from using feature codes..., Carlos Chavez
- Re: Setting a CDR field from using feature codes..., Asghar Mohammad
  - Re: Setting a CDR field from using feature codes..., Carlos Chavez
    - Re: Setting a CDR field from using feature codes..., Asghar Mohammad
      - Re: Setting a CDR field from using feature codes..., Carlos Chavez
        
        Re: Setting a CDR field from using feature codes..., Asghar Mohammad
        
        Re: Setting a CDR field from using feature codes..., Carlos Chavez
        
        Re: Setting a CDR field from using feature codes..., Asghar Mohammad
Follow Me CID, Nyamul Hassan
AMI Reload action, returning generated errors?, John Kiniston
- Re: AMI Reload action, returning generated errors?, Trung Nguyen Dac
ACD problem, Tommy Cooper
- Re: ACD problem, Paul Belanger
- Re: ACD problem, Salman Zafar
  - Re: ACD problem & outbound calls, Tommy Cooper
- Re: ACD problem, Bharat Lalcheta
- Re: ACD problem, Lenz Emilitri
Logging SIP connection status for review, Carlos Alvarez
- Re: Logging SIP connection status for review, Steve Edwards
  - Re: Logging SIP connection status for review, Carlos Alvarez
    - Re: Logging SIP connection status for review, Ishfaq Malik
  - Re: Logging SIP connection status for review, Ron Wheeler
    - Re: Logging SIP connection status for review, Duncan Turnbull
External call control for Asterisk, Simon Green
- Re: External call control for Asterisk, Steve Edwards
  - Re: External call control for Asterisk, Simon Green
- Re: External call control for Asterisk, Lenz Emilitri
  - Re: External call control for Asterisk, Asghar Mohammad
my "blacklist" is not working, Joseph
- Re: my "blacklist" is not working, Joseph
- Re: my "blacklist" is not working, Joseph
  - Re: my "blacklist" is not working, Matthew Jordan
    - Re: my "blacklist" is not working, Joseph
realtime peer w/ callbackextension does not register after 'sip reload', Marie Fischer
Connect to an outbound channel and dial a phone number??, Thomas Perron
- Re: Connect to an outbound channel and dial a phone number??, Marie Fischer
Asterisk Peaking and 91 Calls And not a Dime More!, Nick Khamis
- Re: Asterisk Peaking and 91 Calls And not a Dime More!, Paul Belanger
  - Re: Asterisk Peaking and 91 Calls And not a Dime More!, Nick Khamis
    - Re: Asterisk Peaking and 91 Calls And not a Dime More!, Marie Fischer
      - Re: Asterisk Peaking and 91 Calls And not a Dime More!, Nick Khamis
        
        Re: Asterisk Peaking and 91 Calls And not a Dime More!, Steve Edwards
        
        Re: Asterisk Peaking and 91 Calls And not a Dime More!, Nick Khamis
Re: [OpenSIPS-Users] 404 When BYE initiated by external callee, Nick Khamis
- Re: [OpenSIPS-Users] 404 When BYE initiated by external callee, Joshua Colp
  - Re: [OpenSIPS-Users] 404 When BYE initiated by external callee, Nick Khamis
    - Re: [OpenSIPS-Users] 404 When BYE initiated by external callee, Joshua Colp
      - Re: [OpenSIPS-Users] 404 When BYE initiated by external callee, Nick Khamis
        
        Re: [OpenSIPS-Users] 404 When BYE initiated by external callee, Joshua Colp
        
        Re: [OpenSIPS-Users] 404 When BYE initiated by external callee, Nick Khamis
        
        Re: [OpenSIPS-Users] 404 When BYE initiated by external callee, Nathan Anderson
Feature request: What about a new DB_IFEXISTS function ?, Olivier
- Re: Feature request: What about a new DB_IFEXISTS function ?, Satish Barot
  - Re: Feature request: What about a new DB_IFEXISTS function ?, Olivier
CDR unanswered setting, Shanavaz E A
- Re: CDR unanswered setting, Marie Fischer
dahdi "strange state" error, Greg Woods
OT - How to simulate public IPs for lab testing, Olivier
- Re: OT - How to simulate public IPs for lab testing, Johan Wilfer
  - Re: OT - How to simulate public IPs for lab testing, Olivier
    - Re: OT - How to simulate public IPs for lab testing, Johan Wilfer
extensions.conf / test DID, Thomas Perron
- Re: extensions.conf / test DID, A J Stiles
  - Re: extensions.conf / test DID, Satish Barot
    - Re: extensions.conf / test DID, Doug Lytle
- Re: extensions.conf / test DID, Jacob . E . Miles
- Re: extensions.conf / test DID, Steve Edwards
[Asterisk 1.6] Mysql cdr addon doen't write full channel infomation when disposition is Failed, Trung Nguyen Dac
Broadvoice/MWI?, Nick B
blacklist/V* - using wildcard, Joseph
- Re: blacklist/V* - using wildcard, Joseph
  - Re: blacklist/V* - using wildcard, Doug Lytle
    - Re: blacklist/V* - using wildcard, Joseph
      - Re: blacklist/V* - using wildcard, Doug Lytle
fax - sound/tone - dealing with SPAM, Joseph
- Re: fax - sound/tone - dealing with SPAM, Mitul Limbani
  - Re: fax - sound/tone - dealing with SPAM, Joseph
- Re: fax - sound/tone - dealing with SPAM, Doug Lytle
  - Re: fax - sound/tone - dealing with SPAM, Joseph
    - Re: fax - sound/tone - dealing with SPAM, Doug Lytle
    - Re: fax - sound/tone - dealing with SPAM, Doug Lytle
- Re: fax - sound/tone - dealing with SPAM, Steve Edwards
- Re: fax - sound/tone - dealing with SPAM, Patrick Lists
  - Re: fax - sound/tone - dealing with SPAM, Joseph
ring group failure with "ExtensionState: 4", asterisk
- <Possible follow-ups>
- Re: ring group failure with "ExtensionState: 4", C Goodwin
Asterisk SIP deadlocks - update_provisional_keepalive, Duane Larson
- Re: Asterisk SIP deadlocks - update_provisional_keepalive, Duane Larson
  - Re: Asterisk SIP deadlocks - update_provisional_keepalive, Duane Larson
    - Re: Asterisk SIP deadlocks - update_provisional_keepalive, Jim Lucas
      - Re: Asterisk SIP deadlocks - update_provisional_keepalive, Duane Larson
        
        Re: Asterisk SIP deadlocks - update_provisional_keepalive, Duane Larson
TigerJet 320G Chip / TDM400 Chipset / DAHDI Support, Marshall Henderson
- Re: TigerJet 320G Chip / TDM400 Chipset / DAHDI Support, Tzafrir Cohen
  - Re: TigerJet 320G Chip / TDM400 Chipset / DAHDI Support, Marshall Henderson
    - Re: TigerJet 320G Chip / TDM400 Chipset / DAHDI Support, Patrick Lists
      - Re: TigerJet 320G Chip / TDM400 Chipset / DAHDI Support, Marshall Henderson
        
        Re: TigerJet 320G Chip / TDM400 Chipset / DAHDI Support, Patrick Lists
CLI flood : requested media update control 26, Jonas Kellens
- Re: CLI flood : requested media update control 26, A J Stiles
  - Re: CLI flood : requested media update control 26, Jonas Kellens
    - Re: CLI flood : requested media update control 26, A J Stiles
      - Re: CLI flood : requested media update control 26, Jonas Kellens
        
        Re: CLI flood : requested media update control 26, Matthew Jordan
        
        Re: CLI flood : requested media update control 26, Jonas Kellens
Getting DIALSTATUS via agi, Mike Diehl
FreePBX, Asterisk and Twinkle - Testing a new setup, James B. Byrne
Re: asterisk-users Digest, Vol 104, Issue 53, Kanuvar
SRTP woes, John Cahill
Feature request: Need to INVITE to peer with other domain without peer domain addition, Dmitriy Serov
- Re: Feature request: Need to INVITE to peer with other domain without peer domain addition, Barry Flanagan
  - Re: Feature request: Need to INVITE to peer with other domain without peer domain addition, Dmitriy Serov
    - Re: Feature request: Need to INVITE to peer with other domain without peer domain addition, Paul Belanger
      - Re: Feature request: Need to INVITE to peer with other domain without peer domain addition, Dmitriy Serov
ISDN- E1 PRI module in network side signaling, Dimitar Dimitrov
- Re: ISDN- E1 PRI module in network side signaling, Mitul Limbani
  - Re: ISDN- E1 PRI module in network side signaling, Tony Mountifield
    - Re: ISDN- E1 PRI module in network side signaling, Dimitar Dimitrov
IPv6, Hans Witvliet
- Re: IPv6, Matthew Jordan
Getting Unknown Error while configuring Asterisk with Linux HA, Ahmed Munir
"sip set debug on" output to file only (not to console), Marie Fischer
- Re: "sip set debug on" output to file only (not to console), Doug Lytle
  - Re: "sip set debug on" output to file only (not to console), Marie Fischer
- Re: "sip set debug on" output to file only (not to console), Doug Lytle
- Re: "sip set debug on" output to file only (not to console), Asghar Mohammad
- Re: "sip set debug on" output to file only (not to console), Mitch Claborn
Re: Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved], Olivier
- Re: Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved], Julian Lyndon-Smith
  - Re: Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved], Olivier
    - Re: Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved], Julian Lyndon-Smith
      - Re: Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved], Olivier
Asterisk 11 -CDR values changed in hangup handler not saved ?, Olivier
- Re: Asterisk 11 -CDR values changed in hangup handler not saved ?, Julian Lyndon-Smith
- Re: Asterisk 11 -CDR values changed in hangup handler not saved ?, Mitch Claborn
  - Re: Asterisk 11 -CDR values changed in hangup handler not saved ?, Olivier
    - Re: Asterisk 11 -CDR values changed in hangup handler not saved ?, Mitch Claborn
To queue or not to queue..., Gregory Malsack
- Re: To queue or not to queue..., Carlos Alvarez
  - Re: To queue or not to queue..., Gregory Malsack
- Re: To queue or not to queue..., Chad Wallace
  - Re: To queue or not to queue..., Bharat Lalcheta
- Re: To queue or not to queue..., Lenz Emilitri
Asterisk 11.3.0 Now Available, Asterisk Development Team
Asterisk 1.8.21.0 Now Available, Asterisk Development Team
dahdi-channels.conf vs. chan_dahdi.conf, Ken D'Ambrosio
- Re: dahdi-channels.conf vs. chan_dahdi.conf, Doug Lytle
- Re: dahdi-channels.conf vs. chan_dahdi.conf, Yves A.
- Re: dahdi-channels.conf vs. chan_dahdi.conf, Rusty Newton
chan_mobile: FXS, Hans Witvliet
AST-2013-003: Username disclosure in SIP channel driver, Asterisk Security Team
AST-2013-002: Denial of Service in HTTP server, Asterisk Security Team
AST-2013-001: Buffer Overflow Exploit Through SIP SDP Header, Asterisk Security Team
Asterisk 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones, 11.2.2 Now Available (Security Release), Asterisk Development Team
Pattern matching repeating digits, Nathan Anderson
- Re: Pattern matching repeating digits, Eric Wieling
  - Re: Pattern matching repeating digits, Nathan Anderson
question about zapata.conf, Salaheddine Elharit
- Re: question about zapata.conf, Tzafrir Cohen
  - Re: question about zapata.conf, Salaheddine Elharit
- Re: question about zapata.conf, Yves A.
  - Re: question about zapata.conf, Salaheddine Elharit
    - Re: question about zapata.conf, Eric Wieling
      - Re: question about zapata.conf, Salaheddine Elharit
        
        Re: question about zapata.conf, Tzafrir Cohen
        
        Re: question about zapata.conf, Yves A.
        
        Re: question about zapata.conf, Salaheddine Elharit
Using type=friend a mistake?, Nick Khamis
Asterisk 11, hangup-handlers, Local channels and channel originate, Olivier
- Re: Asterisk 11, hangup-handlers, Local channels and channel originate, Matthew Jordan
  - Re: Asterisk 11, hangup-handlers, Local channels and channel originate, Olivier
    - Re: Asterisk 11, hangup-handlers, Local channels and channel originate, Olivier
      - Re: Asterisk 11, hangup-handlers, Local channels and channel originate, Richard Mudgett
rtcachefriends and rtautoclear on change password, Leandro Dardini
- Re: rtcachefriends and rtautoclear on change password, Michael L. Young
  - Re: rtcachefriends and rtautoclear on change password, Leandro Dardini
Re: Asterisk 11, hangup-handlers, Local channels and channel originate [SOLVED], Olivier
Fundemental changes to CDR within single asterisk family, Ishfaq Malik
- Re: Fundemental changes to CDR within single asterisk family, Matthew Jordan
  - Re: Fundemental changes to CDR within single asterisk family, Ishfaq Malik
  - Re: Fundemental changes to CDR within single asterisk family, Ishfaq Malik
    - Re: Fundemental changes to CDR within single asterisk family, Steve Davies
      - Re: Fundemental changes to CDR within single asterisk family, Ishfaq Malik
Pointer to debug "Got SETUP with duplicate call ptr . Dropping call.", Olivier
- Re: Pointer to debug "Got SETUP with duplicate call ptr . Dropping call.", Richard Mudgett
WARNING[28151] from CLI, Salaheddine Elharit
- Re: WARNING[28151] from CLI, Thorsten Göllner
- Re: WARNING[28151] from CLI, Yves A.
  - Re: WARNING[28151] from CLI, Salaheddine Elharit
    - Re: WARNING[28151] from CLI, Thorsten Göllner
      - Re: WARNING[28151] from CLI, Salaheddine Elharit
        
        Re: WARNING[28151] from CLI, Yves A.
        
        Re: WARNING[28151] from CLI, Salaheddine Elharit
Dedicated LCR Solutions, Nick Khamis
- Re: Dedicated LCR Solutions, Lenz Emilitri
Optimizing Asterisk Environment, Nick Khamis
- Re: Optimizing Asterisk Environment, Leandro Dardini
  - Re: Optimizing Asterisk Environment, Nick Khamis
    - Re: Optimizing Asterisk Environment, Joshua Colp
      - Re: Optimizing Asterisk Environment, Andrew Latham
        
        Re: Optimizing Asterisk Environment, Nick Khamis
        
        Re: Optimizing Asterisk Environment, Andrew Latham

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