asterisk-users
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[asterisk-users] Question on ring count on incoming circuits
Steve Matzura
Re: [asterisk-users] Question on ring count on incoming circuits
Doug Lytle
[asterisk-users] A stupid problem with Playback
Steve Matzura
Re: [asterisk-users] A stupid problem with Playback
asterisk
Re: [asterisk-users] A stupid problem with Playback
Steve Matzura
Re: [asterisk-users] A stupid problem with Playback
asterisk
Re: [asterisk-users] A stupid problem with Playback
Steve Matzura
Re: [asterisk-users] A stupid problem with Playback
asterisk
Re: [asterisk-users] A stupid problem with Playback
Steve Edwards
[asterisk-users] Problems solved
Steve Matzura
Re: [asterisk-users] Problems solved
Sean Bright
Re: [asterisk-users] Problems solved
Steve Matzura
Re: [asterisk-users] Problems solved
asterisk
Re: [asterisk-users] Problems solved
Steve Matzura
[asterisk-users] voip.ms ( was Re: Problems solved )
John Novack
[asterisk-users] Function DENOISE not registered
Fourhundred Thecat
Re: [asterisk-users] Function DENOISE not registered
Doug Lytle
Re: [asterisk-users] Function DENOISE not registered
Fourhundred Thecat
Re: [asterisk-users] Problems Solved, Two Remaining
Steve Matzura
[asterisk-users] Problems Solved, two left
Steve Matzura
Re: [asterisk-users] Problems Solved, two left
Doug Lytle
Re: [asterisk-users] Problems Solved, two left
Stefan Tichy
Re: [asterisk-users] Problems Solved, two left
Stefan Tichy
Re: [asterisk-users] Problems Solved, two left
Doug Lytle
Re: [asterisk-users] Problems Solved, two left
Doug Lytle
Re: [asterisk-users] Problems Solved, two left
Daryl Richards
Re: [asterisk-users] Problems Solved, two left
Steve Matzura
[asterisk-users] Problems with inbound connection and registering phone
Steve Matzura
Re: [asterisk-users] Problems with inbound connection and registering phone
Steve Edwards
Re: [asterisk-users] Problems with inbound connection and registering phone
Steve Edwards
Re: [asterisk-users] Problems with inbound connection and registering phone
Henning Follmann
[asterisk-users] Asterisk Release 20.3.0
Asterisk Development Team
[asterisk-users] Asterisk Release 18.18.0
Asterisk Development Team
[asterisk-users] Ready to throw up my hands in defeat
Steve Matzura
Re: [asterisk-users] Ready to throw up my hands in defeat
TTT
[asterisk-users] FW: Ready to throw up my hands in defeat
TTT
Re: [asterisk-users] FW: Ready to throw up my hands in defeat
Stefan Tichy
Re: [asterisk-users] Ready to throw up my hands in defeat
Steve Matzura
[asterisk-users] SAY_DTMF_INTERRUPT not working
Dovid Bender
Re: [asterisk-users] SAY_DTMF_INTERRUPT not working
Joshua C. Colp
Re: [asterisk-users] SAY_DTMF_INTERRUPT not working
Dovid Bender
[asterisk-users] asterisk 18.17.1 unreachable
Jerry Geis
Re: [asterisk-users] asterisk 18.17.1 unreachable
Antony Stone
[asterisk-users] Calls running forever / CDRs inaccurate
Markus
[asterisk-users] Opus: No translation path after upgrade ubuntu focal => jammy
Benoît Panizzon
Re: [asterisk-users] Opus: No translation path after upgrade ubuntu focal => jammy
Joshua C. Colp
[asterisk-users] DUNDI anyone?
Benoit Panizzon
Re: [asterisk-users] DUNDI anyone?
TTT
[asterisk-users] Broken link in LICENSE file
John Runyon
Re: [asterisk-users] Broken link in LICENSE file
Joshua C. Colp
[asterisk-users] Compiling asterisk makes Systemd timeout when starting the service
Federico
[asterisk-users] Asterisk issue reporting is now live on GitHub
Asterisk Development Team
[asterisk-users] Asterisk translates 200 OK + SDP into 488 not acceptable here after both side agreed on codec.
Benoît Panizzon
Re: [asterisk-users] Asterisk translates 200 OK + SDP into 488 not acceptable here after both side agreed on codec.
Joshua C. Colp
[asterisk-users] Reminder: Issues and Code Contribution move to GitHub
Asterisk Development Team
[asterisk-users] ODBC Crash
Federico
[asterisk-users] Source code for AGI GET DATA command
Rhys Hanrahan
Re: [asterisk-users] Source code for AGI GET DATA command
asterisk
Re: [asterisk-users] Source code for AGI GET DATA command
Rhys Hanrahan
[asterisk-users] Asterisk Infrastructure Move to GitHub
George Joseph
[asterisk-users] couldn't allocate a port for RTP instance
Fourhundred Thecat
Re: [asterisk-users] couldn't allocate a port for RTP instance
Fourhundred Thecat
Re: [asterisk-users] couldn't allocate a port for RTP instance
Joshua C. Colp
Re: [asterisk-users] couldn't allocate a port for RTP instance
Fourhundred Thecat
Re: [asterisk-users] couldn't allocate a port for RTP instance
Joshua C. Colp
[asterisk-users] Setting PJSIP header from AMI
Alex Zarubin
Re: [asterisk-users] Setting PJSIP header from AMI
Joshua C. Colp
[asterisk-users] TLS and NAT
Steve Matzura
Re: [asterisk-users] TLS and NAT
Michael Maier
Re: [asterisk-users] TLS and NAT
Steve Matzura
Re: [asterisk-users] TLS and NAT
Michael Maier
[asterisk-users] Remote-Party-ID set to 0 on re-invite using pjsip in Asterisk 16.
Steve Sether
Re: [asterisk-users] Remote-Party-ID set to 0 on re-invite using pjsip in Asterisk 16.
Steve Sether
[asterisk-users] Intro and question
Steve Matzura
Re: [asterisk-users] Intro and question
Antony Stone
Re: [asterisk-users] Intro and question
Steve Matzura
Re: [asterisk-users] Intro and question
Jeff LaCoursiere
Re: [asterisk-users] Intro and question
Antony Stone
Re: [asterisk-users] Intro and question
Steve Matzura
Re: [asterisk-users] Intro and question
Steve Matzura
Re: [asterisk-users] Intro and question
Frank Vanoni
[asterisk-users] log custom variable in cdr
Fourhundred Thecat
Re: [asterisk-users] log custom variable in cdr
Doug Lytle
[asterisk-users] Asterisk 20.2.1 Now Available
Asterisk Development Team
[asterisk-users] Asterisk 18.17.1 Now Available
Asterisk Development Team
[asterisk-users] 401 error
Jerry Geis
Re: [asterisk-users] 401 error
Steve Edwards
Re: [asterisk-users] 401 error
Joshua C. Colp
Re: [asterisk-users] 401 error
Jerry Geis
Re: [asterisk-users] 401 error
Joshua C. Colp
Re: [asterisk-users] 401 error
Jerry Geis
Re: [asterisk-users] 401 error
Joshua C. Colp
Re: [asterisk-users] 401 error
Jerry Geis
[asterisk-users] Asterisk 20.2.0 Now Available
Asterisk Development Team
[asterisk-users] Asterisk 18.17.0 Now Available
Asterisk Development Team
[asterisk-users] cdr_sqlite3
Fourhundred Thecat
Re: [asterisk-users] cdr_sqlite3
Sean Bright
Re: [asterisk-users] cdr_sqlite3
Steve Edwards
Re: [asterisk-users] cdr_sqlite3
Fourhundred Thecat
Re: [asterisk-users] cdr_sqlite3
Sean Bright
Re: [asterisk-users] cdr_sqlite3
Fourhundred Thecat
[asterisk-users] Mailing Lists
Joshua C. Colp
[asterisk-users] Ping
Benoit Panizzon
[asterisk-users] Asterisk PJSIP setting don't fragment bit on UDP
Benoit Panizzon
[asterisk-users] 5s delays before executing the dialplan
Kingsley Tart
Re: [asterisk-users] 5s delays before executing the dialplan
Joshua C. Colp
Re: [asterisk-users] 5s delays before executing the dialplan
Kingsley Tart
Re: [asterisk-users] 5s delays before executing the dialplan
Joshua C. Colp
[asterisk-users] RTP address learning and timing problem
David Cunningham
Re: [asterisk-users] RTP address learning and timing problem
David Cunningham
Re: [asterisk-users] RTP address learning and timing problem
Joshua C. Colp
Re: [asterisk-users] RTP address learning and timing problem
David Cunningham
Re: [asterisk-users] RTP address learning and timing problem
Joshua C. Colp
Re: [asterisk-users] RTP address learning and timing problem
David Cunningham
Re: [asterisk-users] RTP address learning and timing problem
David Cunningham
Re: [asterisk-users] RTP address learning and timing problem
Joshua C. Colp
Re: [asterisk-users] RTP address learning and timing problem
David Cunningham
Re: [asterisk-users] RTP address learning and timing problem
Joshua C. Colp
Re: [asterisk-users] RTP address learning and timing problem
David Cunningham
[asterisk-users] Asterisk simply stops call processing
Antony Stone
Re: [asterisk-users] Asterisk simply stops call processing
John Harragin
Re: [asterisk-users] Asterisk simply stops call processing
Antony Stone
Re: [asterisk-users] Asterisk simply stops call processing
John Harragin
[asterisk-users] Not reporting IP of the incoming connection 18.14.0
Jerry Geis
[asterisk-users] github - mlan
Jeff LaCoursiere
Re: [asterisk-users] github - mlan
John Runyon
Re: [asterisk-users] github - mlan
Jeff LaCoursiere
[asterisk-users] Asterisk rtp.conf stunaddr setting - what happens if there is an outage
Dan Cropp
Re: [asterisk-users] Asterisk rtp.conf stunaddr setting - what happens if there is an outage
Dan Cropp
Re: [asterisk-users] Asterisk rtp.conf stunaddr setting - what happens if there is an outage
Joshua C. Colp
Re: [asterisk-users] [External] Asterisk rtp.conf stunaddr setting - what happens if there is an outage
Dan Cropp
Re: [asterisk-users] [External] Asterisk rtp.conf stunaddr setting - what happens if there is an outage
Joshua C. Colp
Re: [asterisk-users] [External] [External] Asterisk rtp.conf stunaddr setting - what happens if there is an outage
Dan Cropp
[asterisk-users] set codec based on B side
Fabian Borot
Re: [asterisk-users] set codec based on B side
Joshua C. Colp
[asterisk-users] Is there a list of Channel ARI requests that are allowed when the call is not handed off to the Stasis application
Dan Cropp
Re: [asterisk-users] Is there a list of Channel ARI requests that are allowed when the call is not handed off to the Stasis application
Joshua C. Colp
Re: [asterisk-users] [External] Is there a list of Channel ARI requests that are allowed when the call is not handed off to the Stasis application
Dan Cropp
Re: [asterisk-users] asterisk-users Digest, Vol 221, Issue 2
Ron Lockard
[asterisk-users] Question on ARI externalMedia
Dan Cropp
Re: [asterisk-users] Question on ARI externalMedia
Dan Cropp
[asterisk-users] Testing
Joshua C. Colp
[asterisk-users] Certified Asterisk 18.9-cert4 Now Available
Asterisk Development Team
[asterisk-users] mailing list working?
marek
Re: [asterisk-users] mailing list working?
Antony Stone
Re: [asterisk-users] mailing list working?
Doug Lytle
Re: [asterisk-users] mailing list working?
David Rebarchik
[asterisk-users] sip trunk, parsing DID
Marc SCHAEFER
[asterisk-users] Global variables in global variables
Antony Stone
[asterisk-users] Global variables in global variables
Antony Stone
[asterisk-users] Global variables in global variables
Antony Stone
Re: [asterisk-users] Global variables in global variables
Joel Serrano
Re: [asterisk-users] Global variables in global variables
Antony Stone
[asterisk-users] Global variables in global variables
Antony Stone
Re: [asterisk-users] Global variables in global variables
John Novack
Re: [asterisk-users] Global variables in global variables
Antony Stone
Re: [asterisk-users] Global variables in global variables
Daniel
Re: [asterisk-users] Global variables in global variables
Antony Stone
Re: [asterisk-users] Global variables in global variables
Daniel
Re: [asterisk-users] Global variables in global variables
Antony Stone
Re: [asterisk-users] Global variables in global variables
Sean Bright
Re: [asterisk-users] Global variables in global variables
Antony Stone
[asterisk-users] sender IP of unwanted SIP user
astuserlist
[asterisk-users] Dahdi Compile on 22.04 LTS
Jerry Geis
[asterisk-users] PlayBack
astuserlist
Re: [asterisk-users] PlayBack
Joshua C. Colp
Re: [asterisk-users] PlayBack
astuserlist
Re: [asterisk-users] PlayBack
Joshua C. Colp
Re: [asterisk-users] PlayBack
astuserlist
[asterisk-users] monitor files gsm format split
astuserlist
Re: [asterisk-users] [External] monitor files gsm format split
Dan Cropp
[asterisk-users] Two calls from same server to end device
Jerry Geis
Re: [asterisk-users] Two calls from same server to end device
Jerry Geis
[asterisk-users] Asterisk 18.12.1 to 18.15.0 upgrade seems to have introduced a behavior where PJSIP is unable to send a response to OPTIONS (seems to resolve after anywhere a period of time)
Dan Cropp
Re: [asterisk-users] Asterisk 18.12.1 to 18.15.0 upgrade seems to have introduced a behavior where PJSIP is unable to send a response to OPTIONS (seems to resolve after anywhere a period of time)
Joshua C. Colp
Re: [asterisk-users] Asterisk 18.12.1 to 18.15.0 upgrade seems to have introduced a behavior where PJSIP is unable to send a response to OPTIONS (seems to resolve after anywhere a period of time)
Joshua C. Colp
Re: [asterisk-users] [External] Asterisk 18.12.1 to 18.15.0 upgrade seems to have introduced a behavior where PJSIP is unable to send a response to OPTIONS (seems to resolve after anywhere a period of time)
Dan Cropp
[asterisk-users] Receive DTMF and record audio at the same time
Rhys Hanrahan
Re: [asterisk-users] Receive DTMF and record audio at the same time
asterisk
Re: [asterisk-users] Receive DTMF and record audio at the same time
Rhys Hanrahan
[asterisk-users] cannot load res_geolocation.so
Nick Olsen
Re: [asterisk-users] cannot load res_geolocation.so
Joshua C. Colp
Re: [asterisk-users] cannot load res_geolocation.so
Nick Olsen
Re: [asterisk-users] cannot load res_geolocation.so
John Harragin
[asterisk-users] Codec opus returned invalid number of samples
Fourhundred Thecat
[asterisk-users] Upgraded from asterisk 18.14.0 to 20.0.0 and inbound registration(?) is now failing
Justin Piszcz
Re: [asterisk-users] Upgraded from asterisk 18.14.0 to 20.0.0 and inbound registration(?) is now failing
Mark Murawski
Re: [asterisk-users] Upgraded from asterisk 18.14.0 to 20.0.0 and inbound registration(?) is now failing
John Harragin
[asterisk-users] Asterisk 16.29.1, 18.15.1, 19.7.1, 20.0.1 Now Available
Asterisk Development Team
[asterisk-users] G.729 Annex B or AB support in Asterisk
Tahir Almas Dhesi
[asterisk-users] Asterisk unable to do DNS lookups
TTT
Re: [asterisk-users] Asterisk unable to do DNS lookups
Joshua C. Colp
[asterisk-users] possibility to cancel call duration limit set in app Dial with options S(x) or L(x:y:z) during a call
Nenad Radosavljevic
Re: [asterisk-users] possibility to cancel call duration limit set in app Dial with options S(x) or L(x:y:z) during a call
Joshua C. Colp
Re: [asterisk-users] possibility to cancel call duration limit set in app Dial with options S(x) or L(x:y:z) during a call
Nenad Radosavljevic
[asterisk-users] MixMonitor not recording through transfer
Carlos Chavez
Re: [asterisk-users] MixMonitor not recording through transfer
Joshua C. Colp
[asterisk-users] Handling SIP refers when using a SIP Proxy
Dovid Bender
[asterisk-users] Voicemail Transcription with openai/whisper
Doug Lytle
Re: [asterisk-users] Voicemail Transcription with openai/whisper
Doug Lytle
Earlier messages