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• • Re: [asterisk-users] CLI color prompt Michael Maier
  • Re: [asterisk-users] CLI color prompt Fourhundred Thecat
  • Re: [asterisk-users] CLI color prompt Jeff LaCoursiere
• [asterisk-users] Extracting a SIP Header from a 302 Response Saint Michael
• [asterisk-users] PJSIP John T. Bittner
  • Re: [asterisk-users] PJSIP Joshua C. Colp
• [asterisk-users] Compile dahdi 3.1.0 on CentOS 8 Jerry Geis
  • Re: [asterisk-users] Compile dahdi 3.1.0 on CentOS 8 Jerry Geis
• [asterisk-users] Stir-Shaken clarified Saint Michael
  • Re: [asterisk-users] Stir-Shaken clarified Joshua C. Colp
• [asterisk-users] STIR-Shaken Saint Michael
  • Re: [asterisk-users] STIR-Shaken Steve Edwards
• [asterisk-users] Notification when on the phone Doug Lytle
  • Re: [asterisk-users] Notification when on the phone C.Maj
  • Re: [asterisk-users] Notification when on the phone Doug Lytle
  • Re: [asterisk-users] Notification when on the phone C.Maj
  • Re: [asterisk-users] Notification when on the phone Administrator
  • Re: [asterisk-users] Notification when on the phone Doug Lytle
• [asterisk-users] Stir-Shaken for asterisk Saint Michael
  • Re: [asterisk-users] Stir-Shaken for asterisk Jeff LaCoursiere
  • Re: [asterisk-users] Stir-Shaken for asterisk Jeff LaCoursiere
  • Re: [asterisk-users] Stir-Shaken for asterisk Eric Wieling
  • Re: [asterisk-users] Stir-Shaken for asterisk Sebastian Nielsen
  • Re: [asterisk-users] Stir-Shaken for asterisk Steve Edwards
  • Re: [asterisk-users] Stir-Shaken for asterisk John Kiniston
• [asterisk-users] Is it possible to have a single AMI originate ring multiple contacts? Dan Cropp
  • Re: [asterisk-users] Is it possible to have a single AMI originate ring multiple contacts? Joshua C. Colp
  • Re: [asterisk-users] Is it possible to have a single AMI originate ring multiple contacts? Dan Cropp
• [asterisk-users] Realtime PJSIP RTT Nick Olsen
  • Re: [asterisk-users] Realtime PJSIP RTT Joshua C. Colp
  • Re: [asterisk-users] Realtime PJSIP RTT Nick Olsen
  • Re: [asterisk-users] Realtime PJSIP RTT Joshua C. Colp
• [asterisk-users] SIP/2.0 401 Unauthorized basti
• [asterisk-users] Asterisk and Prometheus Olivier
• [asterisk-users] Asterisk : CDR Analyzer Updated Doug Lytle
  • Re: [asterisk-users] Asterisk : CDR Analyzer Updated Mitul Limbani
  • Re: [asterisk-users] Asterisk : CDR Analyzer Updated Doug Lytle
• [asterisk-users] Asterisk and SIP Proxy on same host = media problem Duncan
• [asterisk-users] rotatestrategy = none not working David Cunningham
  • Re: [asterisk-users] rotatestrategy = none not working Steve Edwards
  • Re: [asterisk-users] rotatestrategy = none not working David Cunningham
  • Re: [asterisk-users] rotatestrategy = none not working Steve Edwards
  • Re: [asterisk-users] rotatestrategy = none not working Steve Edwards
• Re: [asterisk-users] PJSIP sending RTP to private address Saint Michael
• [asterisk-users] Help missing Saint Michael
  • Re: [asterisk-users] Help missing Dovid Bender
• [asterisk-users] PJSIP does not stop sending invites after call is canceled Saint Michael
• [asterisk-users] Meaning of RTT in channelstats Michael Maier
  • Re: [asterisk-users] Meaning of RTT in channelstats Doug Lytle
  • Re: [asterisk-users] Meaning of RTT in channelstats Michael Maier
  • Re: [asterisk-users] Meaning of RTT in channelstats Doug Lytle
  • Re: [asterisk-users] Meaning of RTT in channelstats Joshua C. Colp
  • Re: [asterisk-users] Meaning of RTT in channelstats Michael Maier
  • Re: [asterisk-users] Meaning of RTT in channelstats Joshua C. Colp
• [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY? John Hughes
  • Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY? Michael L. Young
  • Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY? John Hughes
  • Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY? John Hughes
  • Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY? Joshua C. Colp
  • Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY? John Hughes
  • Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY? Richard Mudgett
  • Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY? Richard Mudgett
• [asterisk-users] Tip/Question about encoding temporary data for storage in Asterisk variable to use in AGI Jonathan H
• [asterisk-users] i sided recordings in asterisk 16.10 Israel Gottlieb
  • Re: [asterisk-users] 1 sided recordings in asterisk 16.10 Antony Stone
  • Re: [asterisk-users] i sided recordings in asterisk 16.10 Israel Gottlieb
• [asterisk-users] How to build language selection menu ? Olivier
• [asterisk-users] New RTP engine Saint Michael
  • Re: [asterisk-users] New RTP engine marek
• [asterisk-users] Asterisk versions? Dave Woodfall
  • Re: [asterisk-users] Asterisk versions? Ben Ford
  • Re: [asterisk-users] Asterisk versions? Dave Woodfall
  • Re: [asterisk-users] Asterisk versions? Joshua C. Colp
• [asterisk-users] Asterisk 13.22.0 unstable on Azure Centos 7 & cannot encode .gsm files Stefan Viljoen
• [asterisk-users] Changing ssrc Erol Toker
  • Re: [asterisk-users] Changing ssrc Joshua C. Colp
• [asterisk-users] Better way of streaming radio than "musiconhold" for Asterisk 17.4 ? Jonathan H
  • Re: [asterisk-users] Better way of streaming radio than "musiconhold" for Asterisk 17.4 ? Dan Jenkins
  • Re: [asterisk-users] Better way of streaming radio than "musiconhold" for Asterisk 17.4 ? Jonathan H
  • Re: [asterisk-users] Better way of streaming radio than "musiconhold" for Asterisk 17.4 ? Dan Jenkins
• [asterisk-users] Bare Metal vs Containers/vms Dovid Bender
• [asterisk-users] Length of dial string Paddy Grice
  • Re: [asterisk-users] Length of dial string Dovid Bender
  • Re: [asterisk-users] Length of dial string Paddy Grice
  • Re: [asterisk-users] Length of dial string John Covici
  • Re: [asterisk-users] Length of dial string Floimair Florian
  • Re: [asterisk-users] Length of dial string Paddy Grice
• [asterisk-users] Certified Asterisk 16.8-cert2 Now Available Asterisk Development Team
• [asterisk-users] SIP TLS not working, Asterisk 16.9.0 Karsten Wemheuer
  • Re: [asterisk-users] SIP TLS not working, Asterisk 16.9.0 Stefan Tichy
  • Re: [asterisk-users] SIP TLS not working, Asterisk 16.9.0 Karsten Wemheuer
• [asterisk-users] Asterisk 17.4.0 Now Available Asterisk Development Team
• [asterisk-users] Asterisk 16.10.0 Now Available Asterisk Development Team
• [asterisk-users] Asterisk 13.33.0 Now Available Asterisk Development Team
• [asterisk-users] Certified Asterisk 16.8-cert1 Now Available Asterisk Development Team
• [asterisk-users] max_pseudo_channels Dovid Bender
  • Re: [asterisk-users] max_pseudo_channels Shaun Ruffell
• [asterisk-users] Reload dialplan from bash in strict mode Jöran Vinzens
  • Re: [asterisk-users] Reload dialplan from bash in strict mode Dovid Bender
• [asterisk-users] Advice on building a REST API over ASTDB Olivier
• [asterisk-users] Webrtc and iOS devices Teijo
  • Re: [asterisk-users] Webrtc and iOS devices Dan Jenkins
  • Re: [asterisk-users] Webrtc and iOS devices Teijo
  • Re: [asterisk-users] Webrtc and iOS devices Dan Jenkins
  • Re: [asterisk-users] Webrtc and iOS devices Teijo
• [asterisk-users] Mute conference participants Dovid Bender
  • Re: [asterisk-users] Mute conference participants Doug Lytle
  • Re: [asterisk-users] Mute conference participants Dovid Bender
• [asterisk-users] volume control and setting volume on transfer Iain Mott
  • Re: [asterisk-users] volume control and setting volume on transfer Iain Mott
• [asterisk-users] /outgoing/ .call files and RetryTime problem Dave Woodfall
• [asterisk-users] Troubleshooting load issues Dovid Bender
  • Re: [asterisk-users] Troubleshooting load issues Telium Technical Support
  • Re: [asterisk-users] Troubleshooting load issues Dovid Bender
  • Re: [asterisk-users] Troubleshooting load issues Doug Lytle
  • Re: [asterisk-users] Troubleshooting load issues Telium Technical Support
  • Re: [asterisk-users] Troubleshooting load issues Sean Bright
  • Re: [asterisk-users] Troubleshooting load issues Eric Wieling
  • Re: [asterisk-users] Troubleshooting load issues Dovid Bender
• [asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls Stefan Viljoen
  • Re: [asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls Dovid Bender
  • Re: [asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls Stefan Viljoen
  • Re: [asterisk-users] Asterisk 13.22.0 under very high load conditions - freezes in H exten and blocks new calls Olivier
• [asterisk-users] Dialplan - using multiple AND or OR in set is it possible ? Administrator
  • Re: [asterisk-users] Dialplan - using multiple AND or OR in set is it possible ? Antony Stone
  • Re: [asterisk-users] Dialplan - using multiple AND or OR in set is it possible ? Administrator
  • Re: [asterisk-users] Dialplan - using multiple AND or OR in set is it possible ? John Kiniston
• [asterisk-users] What are "non critical" invites? Stefan Viljoen
• [asterisk-users] Free Persian Asterisk Book and COVID-19 Pandemic Mojtaba
• [asterisk-users] how to make a bug report hw
  • Re: [asterisk-users] how to make a bug report Joshua C. Colp
  • Re: [asterisk-users] how to make a bug report hw
  • Re: [asterisk-users] how to make a bug report Joshua C. Colp
• [asterisk-users] RFC4733 (2833) payload during early audio 183? Benoît Panizzon
  • Re: [asterisk-users] RFC4733 (2833) payload during early audio 183? Joshua C. Colp
• [asterisk-users] Outgoing PJSIP using Kamailio Administrator
  • Re: [asterisk-users] Outgoing PJSIP using Kamailio Joshua C. Colp
  • Re: [asterisk-users] Outgoing PJSIP using Kamailio Administrator
• [asterisk-users] Can't block intrusion D'Arcy Cain
  • Re: [asterisk-users] Can't block intrusion Greg Troxel
  • Re: [asterisk-users] Can't block intrusion D'Arcy Cain
  • Re: [asterisk-users] Can't block intrusion Greg Troxel
  • Re: [asterisk-users] Can't block intrusion Mark Boyce
  • Re: [asterisk-users] Can't block intrusion Larry Moore
  • Re: [asterisk-users] Can't block intrusion D'Arcy Cain
  • Re: [asterisk-users] Can't block intrusion Larry Moore
  • Re: [asterisk-users] Can't block intrusion D'Arcy Cain
  • Re: [asterisk-users] Can't block intrusion Larry Moore
  • Re: [asterisk-users] Can't block intrusion D'Arcy Cain
  • Re: [asterisk-users] Can't block intrusion D'Arcy Cain
• [asterisk-users] Stats on CPU's for Confbridge Dovid Bender
• [asterisk-users] Compile Asterisk without CPU specific extensions/optimizations Telium Technical Support
  • Re: [asterisk-users] Compile Asterisk without CPU specific extensions/optimizations Jean-Denis Girard
• [asterisk-users] AX-1600P FXO port configuration Fernando Pizarro
• [asterisk-users] [Asterisk-video] gstreamer integration Jerry Geis
  • Re: [asterisk-users] [Asterisk-video] gstreamer integration Jerry Geis
  • Re: [asterisk-users] [Asterisk-video] gstreamer integration Shishir Pokharel
  • Re: [asterisk-users] [Asterisk-video] gstreamer integration Jerry Geis
  • Re: [asterisk-users] [Asterisk-video] gstreamer integration Jerry Geis
• [asterisk-users] Faking RTP Dovid Bender
• [asterisk-users] Passing audio and video to linux Jerry Geis
• [asterisk-users] E-Mail notification for each received call Kai Herlemann
  • Re: [asterisk-users] E-Mail notification for each received call Administrator
  • Re: [asterisk-users] E-Mail notification for each received call Kai Herlemann
  • Re: [asterisk-users] E-Mail notification for each received call Administrator
  • Re: [asterisk-users] E-Mail notification for each received call Kai Herlemann
• [asterisk-users] audio problem with asterisk and meetme conference John Covici
  • Re: [asterisk-users] audio problem with asterisk and meetme conference Doug Lytle
  • Re: [asterisk-users] audio problem with asterisk and meetme conference John Covici
  • Re: [asterisk-users] audio problem with asterisk and meetme conference Doug Lytle
  • Re: [asterisk-users] audio problem with asterisk and meetme conference John Covici
  • Re: [asterisk-users] audio problem with asterisk and meetme conference Doug Lytle
  • Re: [asterisk-users] audio problem with asterisk and meetme conference John Covici
• [asterisk-users] Asterisk 17.3: No VoiceMailMain when enabling IMAP and ODBC Olivier
  • [asterisk-users] [SOLVED] Re: Asterisk 17.3: No VoiceMailMain when enabling IMAP and ODBC Olivier
• [asterisk-users] Attempting to get BLF working with linphone John Hughes
  • Re: [asterisk-users] Attempting to get BLF working with linphone Joshua C. Colp
  • Re: [asterisk-users] Attempting to get BLF working with linphone John Hughes
  • Re: [asterisk-users] Attempting to get BLF working with linphone John Hughes
  • Re: [asterisk-users] Attempting to get BLF working with linphone Joshua C. Colp
  • Re: [asterisk-users] Attempting to get BLF working with linphone Olivier
  • Re: [asterisk-users] Attempting to get BLF working with linphone John Hughes
  • Re: [asterisk-users] Attempting to get BLF working with linphone Olivier
  • Re: [asterisk-users] Attempting to get BLF working with linphone Olivier
  • Re: [asterisk-users] Attempting to get BLF working with linphone John Hughes
• [asterisk-users] Old Asterisk forums not working Jonathan H
  • Re: [asterisk-users] Old Asterisk forums not working Joshua C. Colp
  • Re: [asterisk-users] Old Asterisk forums not working Jonathan H
  • Re: [asterisk-users] Old Asterisk forums not working Joshua C. Colp
  • Re: [asterisk-users] Old Asterisk forums not working Joshua C. Colp
  • Re: [asterisk-users] Old Asterisk forums not working Jonathan H
  • Re: [asterisk-users] Old Asterisk forums not working Joshua C. Colp
• [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH John Hughes
  • Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH Joshua C. Colp
  • Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH John Hughes
  • Re: [asterisk-users] SIP/2.0 489 Bad Event in reply to a PUBLISH Social Boh
• [asterisk-users] Max calls per box Dovid Bender
  • Re: [asterisk-users] Max calls per box Dan Jenkins
• [asterisk-users] DAHDI not loading Jerry Geis
  • Re: [asterisk-users] DAHDI not loading Doug Lytle

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