Re: Exceptionally long voice queue length, (continued)

• Re: Exceptionally long voice queue length
  Vik Killa - 01/18/12 06:27 AM

• Re: SIP and NAT best practices since recent changes?
  Bryant Zimmerman - 01/11/12 11:15 AM
• Re: SIP and NAT best practices since recent changes?
  Kevin P. Fleming - 01/11/12 12:59 PM

• Re: Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED]
  Olivier - 01/11/12 08:06 AM
• Re: Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED]
  A J Stiles - 01/11/12 10:15 AM
• Re: Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED]
  Kevin P. Fleming - 01/11/12 10:20 AM
• Re: Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ? [SOLVED]
  Olivier - 01/12/12 01:26 AM

• OT - Which iceweasel plugin to play gsm sound files ?
  Olivier - 01/11/12 07:14 AM

• Iax hold events in AMI 1.1
  Alexandre Rodrigues - 01/11/12 05:03 AM

• SIP and NAT best practices since recent changes?
  Steve Davies - 01/11/12 04:31 AM
• Re: SIP and NAT best practices since recent changes?
  Kevin P. Fleming - 01/11/12 08:44 AM
• Re: SIP and NAT best practices since recent changes?
  Steve Davies - 01/11/12 10:55 AM

• Q: SIPNATtraversal.pdf
  Matthias Apitz - 01/11/12 03:06 AM
• Re: Q: SIPNATtraversal.pdf
  Danny Nicholas - 01/11/12 07:33 AM

• Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
  Olivier - 01/11/12 02:12 AM
• Re: Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
  Benny Amorsen - 01/11/12 05:08 AM
• Re: Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
  Olivier - 01/11/12 07:40 AM
• Re: Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
  Benny Amorsen - 01/11/12 01:04 PM
• Re: Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
  Olivier - 01/12/12 01:25 AM
• Re: Most efficient way to send an HTTP GET from the dialplan with asterisk 1.8 and above) ?
  Jim DeVito - 01/11/12 07:33 AM

• Problems faced in load testing of asterisk
  shalu dhamija - 01/11/12 00:06 AM
• Re: Problems faced in load testing of asterisk
  José Pablo Méndez Soto - 01/11/12 03:01 PM

• Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
  Olivier - 01/10/12 11:41 PM
• Re: Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
  José Pablo Méndez Soto - 01/11/12 02:10 PM
• Re: Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
  Kristijan Vrban - 01/17/12 12:13 PM
• Re: Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
  Olivier - 01/17/12 02:44 PM
• Re: Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
  Tim Nelson - 01/17/12 03:43 PM
• Re: Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
  Tzafrir Cohen - 01/18/12 05:12 AM
• Re: Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
  Tzafrir Cohen - 01/18/12 06:06 AM
• Re: Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
  Steve Underwood - 01/18/12 08:26 AM

• No audio available on SIP/172.16.129.13:5060-00000001??
  shalu dhamija - 01/10/12 09:38 PM
• Re: No audio available on SIP/172.16.129.13:5060-00000001??
  virendra bhati - 01/10/12 10:31 PM
• Re: No audio available on SIP/172.16.129.13:5060-00000001??
  shalu dhamija - 01/11/12 05:29 AM

• Odd DTMF problem when receiving calls
  Christopher David Howie - 01/10/12 04:01 PM
• Re: Odd DTMF problem when receiving calls
  David Backeberg - 01/10/12 06:29 PM
• Re: Odd DTMF problem when receiving calls
  Christopher David Howie - 01/20/12 02:36 PM

• Linux Stun Server
  Bryant Zimmerman - 01/10/12 02:19 PM
• Re: Linux Stun Server
  Danny Nicholas - 01/10/12 02:22 PM

• Hang up phone after declined attended transfer
  Carlos Alvarez - 01/10/12 10:03 AM
• Re: Hang up phone after declined attended transfer
  Danny Nicholas - 01/10/12 10:27 AM
• Re: Hang up phone after declined attended transfer
  Carlos Alvarez - 01/10/12 10:37 AM
• Re: Hang up phone after declined attended transfer
  Ryan Wagoner - 01/10/12 11:59 AM
• Re: Hang up phone after declined attended transfer
  Ryan Wagoner - 01/10/12 12:02 PM
• Re: Hang up phone after declined attended transfer
  Carlos Alvarez - 01/10/12 12:11 PM

• Re: Noise in caller handset when dialing out (with dahdi 2.6.0) [SOLVED]
  Olivier - 01/10/12 09:49 AM

• 44Khz files in Asterisk 10
  Danny Nicholas - 01/09/12 01:00 PM
• Re: 44Khz files in Asterisk 10
  Lefteris Zafiris - 01/09/12 01:33 PM
• Re: 44Khz files in Asterisk 10
  Danny Nicholas - 01/09/12 01:41 PM
• Re: 44Khz files in Asterisk 10
  Lefteris Zafiris - 01/09/12 01:44 PM
• Re: 44Khz files in Asterisk 10
  Kevin P. Fleming - 01/09/12 01:55 PM

• message WARNINGfeatures.c: Failed to play transfer sound! and attended transfer hangs up
  Agustina Berretta - 01/09/12 11:55 AM
• Re: message WARNING[] features.c: Failed to play transfer sound! and attended transfer hangs up
  Richard Mudgett - 01/09/12 01:00 PM

• Noise in caller handset when dialing out (with dahdi 2.6.0)
  Olivier - 01/09/12 11:53 AM
• Re: Noise in caller handset when dialing out (with dahdi 2.6.0)
  Richard Mudgett - 01/09/12 12:48 PM
• Re: Noise in caller handset when dialing out (with dahdi 2.6.0)
  Shaun Ruffell - 01/09/12 12:54 PM
• Re: Noise in caller handset when dialing out (with dahdi 2.6.0)
  Shaun Ruffell - 01/09/12 01:55 PM
• Re: Noise in caller handset when dialing out (with dahdi 2.6.0)
  Olivier - 01/10/12 04:43 AM

• Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?
  Alex Villací­s Lasso - 01/09/12 09:47 AM
• Re: Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?
  Eric Wieling - 01/09/12 09:52 AM
• Re: Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?
  Johann Steinwendtner - 01/09/12 11:47 AM
• Re: Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?
  Richard Mudgett - 01/09/12 12:41 PM
• Re: Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?
  Alex Villací­s Lasso - 01/09/12 01:42 PM
• Re: Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?
  C F - 01/09/12 05:13 PM
• Re: Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?
  Johann Steinwendtner - 01/10/12 03:59 AM

• DEBUG Message
  Elliot Murdock - 01/09/12 03:22 AM

• Cisco AS5300 and Digium g729A codec
  Roi Stork - 01/09/12 01:21 AM
• Re: Cisco AS5300 and Digium g729A codec
  Alex Balashov - 01/09/12 01:35 AM
• Re: Cisco AS5300 and Digium g729A codec
  Roi Stork - 01/09/12 02:20 AM
• Re: Cisco AS5300 and Digium g729A codec
  Roi Stork - 01/09/12 07:31 PM
• Re: Cisco AS5300 and Digium g729A codec
  Roi Stork - 01/10/12 00:53 AM

• Asterisk as register server through OpenSIPS
  Ronald Cepres - 01/09/12 01:04 AM
• Re: Asterisk as register server through OpenSIPS
  Olle E. Johansson - 01/10/12 02:03 AM

• create table in mysql using asterisk
  Eyal - 01/09/12 00:46 AM
• Re: create table in mysql using asterisk
  James Sharp - 01/09/12 00:57 AM
• Re: create table in mysql using asterisk
  Eyal - 01/09/12 01:12 AM
• Re: create table in mysql using asterisk
  Tzafrir Cohen - 01/09/12 01:42 AM
• Re: create table in mysql using asterisk
  Tony Mountifield - 01/09/12 02:42 AM
• Re: create table in mysql using asterisk
  Danny Nicholas - 01/09/12 09:41 AM
• Re: create table in mysql using asterisk
  A J Stiles - 01/09/12 10:05 AM
• Re: create table in mysql using asterisk
  Steve Edwards - 01/09/12 11:08 AM
• Re: create table in mysql using asterisk
  Tony Mountifield - 01/09/12 01:16 PM
• Re: create table in mysql using asterisk
  Tzafrir Cohen - 01/09/12 02:10 PM

• video mail is not store
  Durgesh Mishra - 01/08/12 11:23 PM
• Re: video mail is not store
  Alex Vishnev - 01/09/12 04:49 AM

• Answering call from queue, then put back in queue?
  Todd Routhier - 01/08/12 11:29 AM
• Re: Answering call from queue, then put back in queue?
  Jim Dickenson - 01/08/12 11:44 AM
• Re: Answering call from queue, then put back in queue?
  Todd Routhier - 01/08/12 12:39 PM

• cached VMI on manual voicemail update
  Tzafrir Cohen - 01/08/12 05:32 AM
• Re: cached VMI on manual voicemail update
  Tzafrir Cohen - 01/09/12 02:35 PM

• Couple of questions: SIP ALG, allowguest=no
  Gilles - 01/07/12 06:41 AM
• public ip issue with asterisk cluster
  Matt Hamilton - 01/07/12 06:55 AM

• Asterisk 10.0 & 1.4 - iax codec are not compatible
  Joseph - 01/06/12 11:17 PM
• Re: Asterisk 10.0 & 1.4 - iax codec are not compatible
  Tim Nelson - 01/07/12 07:51 AM
• Re: Asterisk 10.0 & 1.4 - iax codec are not compatible
  Joseph - 01/07/12 09:39 AM
• Re: Asterisk 10.0 & 1.4 - iax codec are not compatible
  Tony Mountifield - 01/07/12 10:15 AM
• Re: [SOLVED] Asterisk 10.0 & 1.4 - iax codec are not compatible
  Joseph - 01/07/12 11:28 AM
• Re: [SOLVED] Asterisk 10.0 & 1.4 - iax codec are not compatible
  Eric Wieling - 01/07/12 11:28 AM
• Re: [SOLVED] Asterisk 10.0 & 1.4 - iax codec are not compatible
  Joseph - 01/07/12 11:37 AM
• Re: [SOLVED] Asterisk 10.0 & 1.4 - iax codec are not compatible
  Eric Wieling - 01/07/12 11:43 AM
• [SOLVED - need explanation] Asterisk 10.0 & 1.4 - iax codec are not compatible
  Joseph - 01/07/12 01:58 PM
• Re: Asterisk 10.0 & 1.4 - iax codec are not compatible
  Andres - 01/07/12 10:21 AM
• Re: Asterisk 10.0 & 1.4 - iax codec are not compatible
  Eric Wieling - 01/07/12 11:21 AM

• best softphone for 2012?
  Tom Poe - 01/06/12 03:01 PM
• Re: best softphone for 2012?
  Ross Cameron - 01/06/12 03:04 PM
• Re: best softphone for 2012?
  Tom Poe - 01/06/12 03:57 PM
• Re: best softphone for 2012?
  Raj Mathur (ààà ààààà) - 01/06/12 07:46 PM
• Re: best softphone for 2012?
  sean darcy - 01/07/12 07:29 AM
• Re: best softphone for 2012?
  Gilles - 01/07/12 07:35 AM
• Re: best softphone for 2012?
  Sean Darcy - 01/07/12 10:38 AM
• Re: best softphone for 2012?
  Gilles - 01/07/12 10:44 AM
• Re: best softphone for 2012?
  Hans Witvliet - 01/07/12 09:19 AM

• Why write your dialplan using Lua?
  José Pablo Méndez Soto - 01/06/12 11:53 AM
• Re: Why write your dialplan using Lua?
  Steve Edwards - 01/06/12 12:13 PM
• Re: Why write your dialplan using Lua?
  José Pablo Méndez Soto - 01/06/12 12:34 PM
• Re: Why write your dialplan using Lua?
  Steve Edwards - 01/06/12 01:57 PM
• Re: Why write your dialplan using Lua?
  Danny Nicholas - 01/06/12 01:59 PM
• Re: Why write your dialplan using Lua?
  Steve Edwards - 01/06/12 02:30 PM
• Re: Why write your dialplan using Lua?
  José Pablo Méndez Soto - 01/06/12 04:07 PM

• Streaming Music to 75 callers ..
  Sriram - 01/06/12 10:57 AM

• Change port from 5060 on Snom phone
  Ishfaq Malik - 01/06/12 09:19 AM
• Re: Change port from 5060 on Snom phone
  José Pablo Méndez Soto - 01/06/12 10:44 AM
• Re: Change port from 5060 on Snom phone
  Paul Hayes - 01/10/12 08:19 AM
• Re: Change port from 5060 on Snom phone
  José Pablo Méndez Soto - 01/10/12 08:28 AM

• Connecting to an Old Phone System
  Dan Journo - 01/06/12 06:15 AM
• Re: Connecting to an Old Phone System
  A J Stiles - 01/06/12 06:48 AM
• Re: Connecting to an Old Phone System
  C F - 01/09/12 05:42 AM
• Re: Connecting to an Old Phone System
  Don Kelly - 01/08/12 10:45 AM
• Re: Connecting to an Old Phone System
  Paul Hayes - 01/10/12 08:17 AM

• no audio using g729A for Cisco AS5300 sip peer
  Roi Stork - 01/06/12 00:35 AM

• calling specific 1800-number not going through.
  Joseph - 01/05/12 06:47 PM
• Re: calling specific 1800-number not going through.
  Jim Dickenson - 01/05/12 06:56 PM
• Re: calling specific 1800-number not going through.
  Joseph - 01/05/12 08:13 PM

• which choice: asterisk-gui or freepbx?
  Tom Poe - 01/05/12 05:56 PM

• asterisk 1.8.8 - caller ID not working.
  Joseph - 01/05/12 04:44 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Joseph - 01/05/12 05:41 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Bruce B - 01/05/12 08:14 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Joseph - 01/05/12 09:26 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Joseph - 01/06/12 01:39 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Danny Nicholas - 01/06/12 01:45 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Joseph - 01/06/12 03:48 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Danny Nicholas - 01/06/12 03:50 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Joseph - 01/06/12 04:12 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Steve Edwards - 01/06/12 04:07 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Eric Wieling - 01/06/12 04:17 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Joseph - 01/06/12 04:36 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Eric Wieling - 01/06/12 05:09 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Joseph - 01/06/12 05:11 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Elliot Murdock - 01/09/12 03:29 AM
• Re: asterisk 1.8.8 - caller ID not working.
  John Novack - 01/05/12 08:43 PM
• Re: asterisk 1.8.8 - caller ID not working.
  Eric Wieling - 01/06/12 06:18 AM
• Re: asterisk 1.8.8 - caller ID not working.
  Bruce B - 01/06/12 08:22 AM
• Re: asterisk 1.8.8 - caller ID not working.
  Eric Wieling - 01/06/12 08:28 AM

• STOP loading extensions.ael
  Joseph - 01/05/12 03:32 PM
• Re: STOP loading extensions.ael
  Steve Edwards - 01/05/12 03:51 PM
• Re: STOP loading extensions.ael
  Danny Nicholas - 01/05/12 03:53 PM
• Re: STOP loading extensions.ael
  Joseph - 01/05/12 04:41 PM

• Blind transfers being cancelled by asterisk & hanging up on remote caller
  Douglas Mortensen - 01/05/12 01:05 PM
• Re: Blind transfers being cancelled by asterisk & hanging up on remote caller
  Luke Hamburg - 01/07/12 03:21 AM
• Re: Blind transfers being cancelled by asterisk & hanging up on remote caller
  Ryan Wagoner - 01/07/12 09:01 AM
• Re: Blind transfers being cancelled by asterisk & hanging up on remote caller
  Douglas Mortensen - 01/07/12 01:16 PM
• Re: Blind transfers being cancelled by asterisk & hanging up on remote caller
  Douglas Mortensen - 01/07/12 01:16 PM
• Re: Blind transfers being cancelled by asterisk & hanging up on remote caller
  Luke Hamburg - 01/08/12 02:04 PM

• question on CDR
  Jerry Geis - 01/05/12 11:23 AM
• Re: question on CDR
  Danny Nicholas - 01/05/12 12:34 PM

• Best non polycom SIP conference room phone
  Bryant Zimmerman - 01/05/12 10:25 AM
• Re: Best non polycom SIP conference room phone
  José Pablo Méndez Soto - 01/05/12 11:11 AM
• Re: Best non polycom SIP conference room phone
  Carlos Alvarez - 01/05/12 11:12 AM
• Re: Best non polycom SIP conference room phone
  Jamie A. Stapleton - 01/05/12 03:21 PM
• Re: Best non polycom SIP conference room phone
  Kevin P. Fleming - 01/05/12 03:25 PM
• Re: Best non polycom SIP conference room phone
  Leif Madsen - 01/09/12 05:38 AM
• Re: Best non polycom SIP conference room phone
  C F - 01/05/12 05:12 PM
• Re: Best non polycom SIP conference room phone
  Carlos Alvarez - 01/05/12 05:22 PM
• Re: Best non polycom SIP conference room phone
  C F - 01/09/12 04:54 AM
• Re: Best non polycom SIP conference room phone
  BryantZ - 01/08/12 10:05 AM
• Re: Best non polycom SIP conference room phone
  Ryan Wagoner - 01/08/12 01:24 PM
• Re: Best non polycom SIP conference room phone
  Ishfaq Malik - 01/09/12 02:04 AM
• Re: Best non polycom SIP conference room phone
  C F - 01/09/12 05:02 PM
• Re: Best non polycom SIP conference room phone
  Jamie A. Stapleton - 01/10/12 03:28 PM
• Re: Best non polycom SIP conference room phone
  Carlos Alvarez - 01/08/12 01:39 PM
• Re: Best non polycom SIP conference room phone
  Luke Hamburg - 01/08/12 01:59 PM
• Re: Best non polycom SIP conference room phone
  Carlos Alvarez - 01/08/12 02:07 PM
• Re: Best non polycom SIP conference room phone
  Benny Amorsen - 01/09/12 02:27 PM
• Re: Best non polycom SIP conference room phone
  Jim DeVito - 01/09/12 02:38 PM
• Re: Best non polycom SIP conference room phone
  C F - 01/09/12 05:10 PM
• Re: Best non polycom SIP conference room phone
  Bryant Zimmerman - 01/10/12 03:59 PM

• Re: asterisk -> AGI (perl) -> sqlplus(oracle)
  Ahmed Munir - 01/05/12 07:55 AM
• Re: asterisk -> AGI (perl) -> sqlplus(oracle)
  Danny Nicholas - 01/05/12 08:17 AM
• Re: asterisk -> AGI (perl) -> sqlplus(oracle)
  Ahmed Munir - 01/06/12 08:33 AM
• Re: asterisk -> AGI (perl) -> sqlplus(oracle)
  Danny Nicholas - 01/06/12 09:01 AM
• Re: asterisk -> AGI (perl) -> sqlplus(oracle)
  Ron Bergin - 01/06/12 09:31 AM
• Re: asterisk -> AGI (perl) -> sqlplus(oracle)
  Dale Noll - 01/06/12 03:28 PM
• Re: asterisk -> AGI (perl) -> sqlplus(oracle)
  Steve Edwards - 01/07/12 10:43 AM

• Asterisk1.8 support video trancoding ?
  Durgesh Mishra - 01/05/12 03:13 AM
• Re: Asterisk1.8 support video trancoding ?
  Paul Belanger - 01/05/12 07:48 AM
• Re: Asterisk1.8 support video trancoding ?
  Kevin P. Fleming - 01/05/12 09:03 AM

• Video trancoding not done.
  Durgesh Mishra - 01/05/12 02:39 AM
• Re: Video trancoding not done.
  Kevin P. Fleming - 01/05/12 09:02 AM

• Where are the fax instructions?
  José Pablo Méndez Soto - 01/05/12 00:04 AM
• Re: Where are the fax instructions?
  Kevin P. Fleming - 01/05/12 09:05 AM
• Re: Where are the fax instructions?
  José Pablo Méndez Soto - 01/05/12 10:59 AM
• Re: Where are the fax instructions?
  José Pablo Méndez Soto - 01/06/12 11:55 AM
• Re: Where are the fax instructions?
  Kevin P. Fleming - 01/06/12 12:00 PM
• Re: Where are the fax instructions?
  José Pablo Méndez Soto - 01/06/12 12:04 PM
• Re: Where are the fax instructions?
  Kevin P. Fleming - 01/06/12 12:08 PM

• Re: DIALSTATUS Values
  Kamlesh Kumar - 01/04/12 10:41 PM
• Re: DIALSTATUS Values
  Zohair Raza - 01/04/12 11:53 PM

• Re: Server-to-server BLF
  Ronald Cepres - 01/04/12 09:08 PM
• Re: Server-to-server BLF
  Ishfaq Malik - 01/06/12 02:16 AM
• Re: Server-to-server BLF
  Ronald Cepres - 01/13/12 00:01 AM
• Re: Server-to-server BLF
  Leandro Dardini - 01/13/12 00:45 AM
• Re: Server-to-server BLF
  Ronald Cepres - 01/16/12 03:25 AM

• DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0 Released
  Asterisk Development Team - 01/04/12 02:53 PM
• Re: DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0 Released
  David Backeberg - 01/05/12 06:33 PM

• question sangoma vs digium
  Agustina Berretta - 01/04/12 01:48 PM
• Re: question sangoma vs digium
  James zhu - 01/04/12 06:22 PM

• From address missing 'sip:', using it anyway
  motty.cruz - 01/04/12 12:45 PM
• Re: From address missing 'sip:', using it anyway
  Sammy Govind - 01/04/12 11:51 PM

• asterisk -> AGI (perl) -> sqlplus (oracle)
  Ahmed Munir - 01/04/12 09:46 AM
• Re: asterisk -> AGI (perl) -> sqlplus (oracle)
  Danny Nicholas - 01/04/12 10:36 AM
• Re: asterisk -> AGI (perl) -> sqlplus (oracle)
  Ahmed Munir - 01/04/12 11:17 AM
• Re: asterisk -> AGI (perl) -> sqlplus (oracle)
  Danny Nicholas - 01/04/12 11:21 AM
• Re: asterisk -> AGI (perl) -> sqlplus (oracle)
  Steve Edwards - 01/04/12 12:20 PM
• Re: asterisk -> AGI (perl) -> sqlplus (oracle)
  LL - 01/05/12 00:30 AM
• Re: asterisk -> AGI (perl) -> sqlplus (oracle)
  Ron Bergin - 01/05/12 03:10 PM

• Which QSIG variant and profiles does asterisk support ?
  Olivier - 01/04/12 09:21 AM

• Set Call Codec in extension.conf
  Faraj Khasib - 01/04/12 04:58 AM
• Re: Set Call Codec in extension.conf
  Faraj Khasib - 01/04/12 10:11 AM
• Re: Set Call Codec in extension.conf
  Eric Wieling - 01/04/12 10:12 AM
• Re: Set Call Codec in extension.conf
  Faraj Khasib - 01/04/12 10:16 AM
• Re: Set Call Codec in extension.conf
  Eric Wieling - 01/04/12 10:20 AM
• Re: Set Call Codec in extension.conf
  Faraj Khasib - 01/04/12 10:28 AM
• Re: Set Call Codec in extension.conf
  Danny Nicholas - 01/04/12 10:30 AM
• Re: Set Call Codec in extension.conf
  Faraj Khasib - 01/04/12 10:36 AM
• Re: Set Call Codec in extension.conf
  Danny Nicholas - 01/04/12 10:41 AM
• Re: Set Call Codec in extension.conf
  Faraj Khasib - 01/04/12 10:45 AM
• Re: Set Call Codec in extension.conf
  Danny Nicholas - 01/04/12 10:47 AM
• Re: Set Call Codec in extension.conf
  Faraj Khasib - 01/04/12 10:52 AM
• Re: Set Call Codec in extension.conf
  Danny Nicholas - 01/04/12 10:56 AM
• Re: Set Call Codec in extension.conf
  Faraj Khasib - 01/04/12 10:58 AM
• Re: Set Call Codec in extension.conf
  Faraj Khasib - 01/04/12 11:00 AM
• Re: Set Call Codec in extension.conf
  Danny Nicholas - 01/04/12 11:03 AM
• Re: Set Call Codec in extension.conf
  Faraj Khasib - 01/04/12 11:07 AM
• Re: Set Call Codec in extension.conf
  Danny Nicholas - 01/04/12 11:11 AM
• Re: Set Call Codec in extension.conf
  Faraj Khasib - 01/04/12 11:14 AM
• Re: Set Call Codec in extension.conf
  Danny Nicholas - 01/04/12 11:17 AM
• Re: Set Call Codec in extension.conf
  Faraj Khasib - 01/04/12 11:17 AM

• Asterisk won't start - trap invalid opcode
  Duncan Turnbull - 01/04/12 03:15 AM
• Re: Asterisk won't start - trap invalid opcode
  A J Stiles - 01/04/12 03:49 AM
• Re: Asterisk won't start - trap invalid opcode
  Duncan Turnbull - 01/04/12 04:15 AM
• Re: Asterisk won't start - trap invalid opcode
  Duncan Turnbull - 01/04/12 06:22 AM
• Re: Asterisk won't start - trap invalid opcode
  A J Stiles - 01/04/12 11:11 AM
• Re: Asterisk won't start - trap invalid opcode
  Steve Edwards - 01/04/12 12:08 PM
• Re: Asterisk won't start - trap invalid opcode
  Duncan Turnbull - 01/04/12 01:35 PM
• Re: Asterisk won't start - trap invalid opcode
  James Cloos - 01/04/12 04:23 PM
• Re: Asterisk won't start - trap invalid opcode
  Duncan Turnbull - 01/04/12 05:29 PM

• Re: Failed to authenticate on INVITE to Anonymous
  Jayesh Labade - 01/04/12 02:39 AM
• Re: Failed to authenticate on INVITE to Anonymous
  virendra bhati - 01/04/12 02:48 AM
• Re: Failed to authenticate on INVITE to Anonymous
  Jayesh Labade - 01/04/12 03:05 AM
• Re: Failed to authenticate on INVITE to Anonymous
  virendra bhati - 01/04/12 04:07 AM
• Re: Failed to authenticate on INVITE to Anonymous
  Jayesh Labade - 01/04/12 04:16 AM
• Re: Failed to authenticate on INVITE to Anonymous
  sean darcy - 01/04/12 12:56 PM

• Rami
  Arjan Kroon | Mobillion - 01/04/12 02:21 AM
• Re: Rami
  gokulnath - 01/04/12 09:33 AM
• Re: Rami
  Arjan Kroon | Mobillion - 01/04/12 11:11 AM
• Re: Rami
  Steve Edwards - 01/04/12 12:08 PM

• Anyone have a reliable T.38 Solution
  Matt Darnell - 01/04/12 01:19 AM
• Re: Anyone have a reliable T.38 Solution
  David Klaverstyn - 01/04/12 04:04 AM
• Re: Anyone have a reliable T.38 Solution
  Matt Darnell - 01/04/12 08:57 PM
• Re: Anyone have a reliable T.38 Solution
  Klaverstyn, David C - 01/04/12 08:58 PM
• Re: Anyone have a reliable T.38 Solution
  Michael Keuter - 01/05/12 04:47 AM
• Re: Anyone have a reliable T.38 Solution
  Steve Underwood - 01/05/12 06:09 AM
• Re: Anyone have a reliable T.38 Solution
  David Backeberg - 01/05/12 05:16 PM
• Re: Anyone have a reliable T.38 Solution
  Benny Amorsen - 01/06/12 08:05 AM
• Re: Anyone have a reliable T.38 Solution
  Ryan Wagoner - 01/06/12 02:50 PM
• Re: Anyone have a reliable T.38 Solution
  Kevin P. Fleming - 01/04/12 04:23 PM
• Re: Anyone have a reliable T.38 Solution
  Olivier - 01/11/12 00:04 AM
• Re: Anyone have a reliable T.38 Solution
  Steve Underwood - 01/11/12 05:52 AM
• Re: Anyone have a reliable T.38 Solution
  Olivier - 01/11/12 08:18 AM
• Re: Anyone have a reliable T.38 Solution
  Kevin P. Fleming - 01/11/12 08:31 AM
• Re: Anyone have a reliable T.38 Solution
  Steve Underwood - 01/11/12 09:42 AM
• Re: Anyone have a reliable T.38 Solution
  Olivier - 01/12/12 02:42 AM
• Re: Anyone have a reliable T.38 Solution
  Frank Sautter - 01/05/12 03:05 AM

• ConfBridge no audio problem
  Roi Stork - 01/03/12 09:16 PM
• Re: ConfBridge no audio problem
  Kevin P. Fleming - 01/12/12 03:58 PM

• Mark queue agent as away
  Raj Mathur (ààà ààààà) - 01/03/12 07:44 PM
• Re: Mark queue agent as away
  James Sharp - 01/03/12 09:41 PM

• Speech recognition in asterisk using google voice API
  Lefteris Zafiris - 01/03/12 07:44 PM
• Re: Speech recognition in asterisk using google voice API
  Bruce B - 01/03/12 10:46 PM
• Re: Speech recognition in asterisk using google voice API
  Bruce B - 01/03/12 10:53 PM
• Re: Speech recognition in asterisk using google voice API
  LL - 01/03/12 11:08 PM
• Re: Speech recognition in asterisk using google voice API
  Lefteris Zafiris - 01/04/12 02:10 AM
• Re: Speech recognition in asterisk using google voice API
  Julian Lyndon-Smith - 01/04/12 07:09 AM
• Re: Speech recognition in asterisk using google voice API
  Lefteris Zafiris - 01/04/12 07:19 AM
• Re: Speech recognition in asterisk using google voice API
  Julian Lyndon-Smith - 01/04/12 07:25 AM
• Re: Speech recognition in asterisk using google voice API
  Lefteris Zafiris - 01/04/12 07:31 AM
• Re: Speech recognition in asterisk using google voice API
  Julian Lyndon-Smith - 01/04/12 07:36 AM
• Re: Speech recognition in asterisk using google voice API
  Bruce B - 01/04/12 11:27 AM
• Re: Speech recognition in asterisk using google voice API
  isrlgb - 01/04/12 11:28 AM
• Re: Speech recognition in asterisk using google voice API
  Lefteris Zafiris - 01/04/12 12:17 PM
• Re: Speech recognition in asterisk using google voice API
  Michelle Dupuis - 01/04/12 11:51 AM
• Re: Speech recognition in asterisk using google voice API
  Lefteris Zafiris - 01/04/12 12:15 PM
• Re: Speech recognition in asterisk using google voice API
  Lefteris Zafiris - 01/04/12 12:27 PM
• Re: Speech recognition in asterisk using google voice API
  sean darcy - 01/04/12 12:50 PM
• Re: Speech recognition in asterisk using google voice API
  Israel Gottlieb - 01/04/12 12:51 PM
• Re: Speech recognition in asterisk using google voice API
  Lefteris Zafiris - 01/04/12 01:00 PM
• Re: Speech recognition in asterisk using google voice API
  Lefteris Zafiris - 01/04/12 04:43 PM
• Re: Speech recognition in asterisk using google voice API
  Bruce B - 01/06/12 06:48 PM
• Re: Speech recognition in asterisk using google voice API
  Lefteris Zafiris - 01/06/12 07:42 PM
• Re: Speech recognition in asterisk using google voice API
  Bruce B - 01/06/12 08:52 PM
• Re: Speech recognition in asterisk using google voice API
  Bruce B - 01/06/12 09:04 PM
• Re: Speech recognition in asterisk using google voice API
  Bruce B - 01/07/12 00:36 AM
• Re: Speech recognition in asterisk using google voice API
  Lefteris Zafiris - 01/07/12 05:24 AM
• Re: Speech recognition in asterisk using google voice API
  Danny Nicholas - 01/12/12 08:54 AM
• Re: Speech recognition in asterisk using google voice API
  Lefteris Zafiris - 01/12/12 12:51 PM

• ISDN E1, after electrical disconnected is not becoming UP, IRQ misses: 1
  bilal ghayyad - 01/03/12 03:43 PM
• Re: ISDN E1, after electrical disconnected is not becoming UP, IRQ misses: 1
  Shaun Ruffell - 01/03/12 04:08 PM

• Layer2 Down in BRI connection
  khalid touati - 01/03/12 12:31 PM
• Layer2 Down in BRI connection
  khalid touati - 01/03/12 04:48 PM

• Ringing agents cell as an alert?
  Todd Routhier - 01/03/12 11:08 AM
• Re: Ringing agents cell as an alert?
  James Sharp - 01/03/12 11:16 AM
• Re: Ringing agents cell as an alert?
  Todd Routhier - 01/03/12 11:23 AM
• Re: Ringing agents cell as an alert?
  James Sharp - 01/03/12 11:33 AM

• Problem connecting to 4569/UDP
  kazabe - 01/03/12 10:56 AM
• Re: Problem connecting to 4569/UDP
  Roland - 01/06/12 04:07 AM
• Re: Problem connecting to 4569/UDP
  kazabe - 01/06/12 09:15 AM
• Re: Problem connecting to 4569/UDP
  Danny Nicholas - 01/06/12 09:17 AM

• NAT/IPTABLES workarounds
  Danny Nicholas - 01/03/12 08:25 AM
• Re: NAT/IPTABLES workarounds
  Patrick Lists - 01/03/12 09:05 AM
• Re: NAT/IPTABLES workarounds
  Kevin P. Fleming - 01/04/12 04:23 PM
• Re: NAT/IPTABLES workarounds
  Sammy Govind - 01/04/12 11:53 PM

• Registering multi-clients
  Faraj Khasib - 01/03/12 07:23 AM

• Question on system command 1.4.43
  Jerry Geis - 01/03/12 06:49 AM
• Re: Question on system command 1.4.43
  Doug Lytle - 01/03/12 07:30 AM
• Re: Question on system command 1.4.43
  Jim DeVito - 01/03/12 07:30 AM
• Re: Question on system command 1.4.43
  Jerry Geis - 01/03/12 11:15 AM
• Re: Question on system command 1.4.43
  Steve Edwards - 01/03/12 11:28 AM
• Re: Question on system command 1.4.43
  Eric Wieling - 01/03/12 11:33 AM

• Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
  Kaushal Shriyan - 01/03/12 06:49 AM
• Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
  Zohair Raza - 01/03/12 06:55 AM
• Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
  Kaushal Shriyan - 01/03/12 06:59 AM
• Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
  Zohair Raza - 01/03/12 07:04 AM
• Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
  Patrick Lists - 01/03/12 07:59 AM

• Using Asterisk as a softphone
  Christian Jaeger - 01/03/12 06:36 AM
• Re: Using Asterisk as a softphone
  Sammy Govind - 01/04/12 10:41 PM

• Re: Issue with dahdi 2.5.0 and Digium HA8-B400M
  Olivier - 01/03/12 04:12 AM

• How to make SIP guest call
  Faraj Khasib - 01/03/12 04:10 AM
• Re: How to make SIP guest call
  Faraj Khasib - 01/03/12 04:35 AM
• Re: How to make SIP guest call
  David Klaverstyn - 01/03/12 04:47 AM
• Re: How to make SIP guest call
  Faraj Khasib - 01/03/12 04:51 AM
• Re: How to make SIP guest call
  Faraj Khasib - 01/03/12 04:58 AM
• Re: How to make SIP guest call
  Faraj Khasib - 01/03/12 06:15 AM
• Re: How to make SIP guest call
  Patrick Lists - 01/03/12 07:56 AM
• Re: How to make SIP guest call
  Faraj Khasib - 01/03/12 08:04 AM
• Re: How to make SIP guest call
  Roland - 01/03/12 08:33 AM
• Re: How to make SIP guest call
  Faraj Khasib - 01/03/12 08:34 AM

• Asterisk 1.8 - BRI D Channel going up and down every few seconds
  Marco Mooijekind - 01/03/12 03:23 AM
• Re: Asterisk 1.8 - BRI D Channel going up and down every few seconds
  Patrick Lists - 01/03/12 04:08 AM
• Re: Asterisk 1.8 - BRI D Channel going up and down every few seconds
  Olivier - 01/03/12 05:22 AM

• dialplan -> dial command -> custom ringtone
  Qqblog Qqblog - 01/03/12 02:09 AM
• Re: dialplan -> dial command -> custom ringtone
  Carlos Rojas - 01/03/12 05:43 AM
• åèï dialplan -> dial command -> custom ringtone
  Qqblog Qqblog - 01/04/12 07:42 PM

• Re: MFCR2 Long distance calls not connected
  Moises Silva - 01/02/12 09:27 PM

• Problem w/ PC port on Polycom 335
  Mike Diehl - 01/02/12 05:39 PM
• Re: Problem w/ PC port on Polycom 335
  Doug Lytle - 01/02/12 05:49 PM
• Re: Problem w/ PC port on Polycom 335
  Jim DeVito - 01/02/12 07:17 PM
• Re: Problem w/ PC port on Polycom 335
  Mike Diehl - 01/04/12 10:05 AM

• Wired attack on Asterisk - Can anyone explain this?
  Bruce B - 01/02/12 01:01 PM

• Help_video voice mail not retriev properly
  Durgesh Mishra - 01/02/12 04:40 AM

• Set Call type in dial plan
  Faraj Khasib - 01/02/12 02:10 AM
• Re: Set Call type in dial plan
  Faraj Khasib - 01/02/12 01:32 PM
• Re: Set Call type in dial plan
  Doug Lytle - 01/02/12 02:50 PM
• Re: Set Call type in dial plan
  Faraj Khasib - 01/02/12 09:29 PM
• Re: Set Call type in dial plan
  virendra bhati - 01/02/12 10:31 PM
• Re: Set Call type in dial plan
  Faraj Khasib - 01/02/12 11:01 PM
• Re: Set Call type in dial plan
  virendra bhati - 01/02/12 11:20 PM
• Re: Set Call type in dial plan
  Faraj Khasib - 01/02/12 11:23 PM
• Re: Set Call type in dial plan
  virendra bhati - 01/02/12 11:31 PM
• Re: Set Call type in dial plan
  Faraj Khasib - 01/03/12 01:58 AM
• Re: Set Call type in dial plan
  Sammy Govind - 01/03/12 02:29 AM
• Re: Set Call type in dial plan
  Faraj Khasib - 01/03/12 02:33 AM
• Re: Set Call type in dial plan
  Sammy Govind - 01/04/12 10:34 PM
• Re: Set Call type in dial plan
  Faraj Khasib - 01/06/12 05:01 AM

• Re: performance/memory
  virendra bhati - 01/01/12 10:44 PM

• tcp version of toronto - osaka doesn't work
  sean darcy - 01/01/12 09:36 PM
• Re: tcp version of toronto - osaka doesn't work
  sean darcy - 01/02/12 09:24 AM
• Re: tcp version of toronto - osaka doesn't work
  sean darcy - 01/02/12 09:31 AM
• Re: tcp version of toronto - osaka doesn't work
  sean darcy - 01/02/12 02:41 PM
• Re: tcp version of toronto - osaka doesn't work
  Leif Madsen - 01/09/12 05:36 AM

• Re: Client - registers but unreachable
  white hat - 01/01/12 05:33 PM
• Re: Client - registers but unreachable
  white hat - 01/01/12 05:43 PM

• registration not authorized - stale nonce
  Matt Hamilton - 01/01/12 04:14 PM
• UPDATE RE: registration not authorized - stale nonce
  Matt Hamilton - 01/01/12 08:42 PM
• Re: registration not authorized - stale nonce
  asterisk jobs - 01/02/12 01:41 PM

• asterisk 1.8 codec negotiation
  covici - 01/01/12 03:18 PM
• Re: asterisk 1.8 codec negotiation
  José Pablo Méndez Soto - 01/01/12 10:17 PM
• Re: asterisk 1.8 codec negotiation
  Kevin P. Fleming - 01/04/12 04:16 PM
• Re: asterisk 1.8 codec negotiation
  covici - 01/04/12 06:08 PM
• Re: asterisk 1.8 codec negotiation
  Kevin P. Fleming - 01/05/12 09:08 AM

• GoAutoDialer, ViciDial and Vicidial group
  bilal ghayyad - 01/01/12 03:15 PM

• Re: 1.6 and 1.8
  Elliot Murdock - 01/01/12 05:26 AM
• Re: 1.6 and 1.8
  mattias - 01/01/12 05:44 AM

• 481 Call leg/transaction does not exists Status Response
  Elliot Murdock - 01/01/12 05:24 AM