November 2019 Archives by author

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Starting: Fri Nov 1 05:18:34 CDT 2019
Ending: Fri Nov 29 09:30:42 CST 2019
Messages: 94

• [asterisk-users] INFO to rfc2833 issues   Dovid Bender
• [asterisk-users] Digium's Opus Codec download links broken?   Dovid Bender
• [asterisk-users] Low cost routing   Dovid Bender
• [asterisk-users] Dropping calls based on IP in SDP   Dovid Bender
• [asterisk-users] multiple softphone clients and same/different account credentials   C.Maj
• [asterisk-users] multiple softphone clients and same/different account credentials   C.Maj
• [asterisk-users] Low cost routing   Olivier CALVANO
• [asterisk-users] Faxes stopped working - AMI issue?   D'Arcy Cain
• [asterisk-users] Stuck "channel"   Carlos Chavez
• [asterisk-users] sip.conf host!=dynamic peer specific options (e.g. directmedia=off, transport=tcp) not working!?   Joshua C. Colp
• [asterisk-users] Digium's Opus Codec download links broken?   Joshua C. Colp
• [asterisk-users] Asterisk 16.6.1: PJSIP: delayed action of core since update to 16.6.1   Joshua C. Colp
• [asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer   Joshua C. Colp
• [asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer   Joshua C. Colp
• [asterisk-users] Global number rewriting rules affecting ALL headers?   Joshua C. Colp
• [asterisk-users] Global number rewriting rules affecting ALL headers?   Joshua C. Colp
• [asterisk-users] trust_id_inbound=yes but take CallerID(Num) from From: not from PAI   Joshua C. Colp
• [asterisk-users] Asterisk 16.6.1: PJSIP: delayed action of core since update to 16.6.1   Joshua C. Colp
• [asterisk-users] Avoid transcoding if both ends support same coded   Joshua C. Colp
• [asterisk-users] PJSIP device_state_busy_at, how does this work?   Joshua C. Colp
• [asterisk-users] PJSIP device_state_busy_at, how does this work?   Joshua C. Colp
• [asterisk-users] Is it possible to record 2-4 party call audio in stereo quality as opposed to mono?   Dan Cropp
• [asterisk-users] Is it possible to record 2-4 party call audio in stereo quality as opposed to mono?   Dan Cropp
• [asterisk-users] Does Asterisk support one-legged transfers with external switches?   Dan Cropp
• [asterisk-users] Digium's Opus Codec download links broken?   Floimair Florian
• [asterisk-users] Digium's Opus Codec download links broken?   Floimair Florian
• [asterisk-users] multiple softphone clients and same/different account credentials   Ben Ford
• [asterisk-users] SIP messaging   Jerry Geis
• [asterisk-users] CDR extract call numbers on interval on unique callers   Andre Gronwald
• [asterisk-users] CDR extract call numbers on interval on unique callers   Andre Gronwald
• [asterisk-users] CDR extract call numbers on interval on unique callers   Andre Gronwald
• [asterisk-users] Simple, fast single-word offline free speech recognition in Asterisk (or as an AGI)?   Jonathan H
• [asterisk-users] Asterisk 16.6.1: PJSIP: delayed action of core since update to 16.6.1   O. Hartmann
• [asterisk-users] Simple, fast single-word offline free speech recognition in Asterisk (or as an AGI)?   Doug Lytle
• [asterisk-users] Own MOH incorrectly kicking in instead of the MOH of the callee   Michael Maier
• [asterisk-users] possible bug in Asterisk 16   Tony Mountifield
• [asterisk-users] pre-dial handler, how to access variables from calling channel?   Tony Mountifield
• [asterisk-users] Stuck "channel"   Richard Mudgett
• [asterisk-users] How to set http.conf for HTTPS support on Debian Buster ?   Richard Mudgett
• [asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer   Sebastian Nielsen
• [asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer   Sebastian Nielsen
• [asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer   Sebastian Nielsen
• [asterisk-users] On Register, run a script, validate source IP   Sebastian Nielsen
• [asterisk-users] What is Debian Buster's asterisk equivalent of Digium's contrib/scripts/ast_tls_cert ?   Olivier
• [asterisk-users] How to set http.conf for HTTPS support on Debian Buster ?   Olivier
• [asterisk-users] Question on WebRTC configuration   Olivier
• [asterisk-users] How to set http.conf for HTTPS support on Debian Buster ?   Olivier
• [asterisk-users] How to set http.conf for HTTPS support on Debian Buster ?   Olivier
• [asterisk-users] WebRTC: which ACL and modes for /etc/asterisk/keys/asterisk.pem ?   Olivier
• [asterisk-users] Which architecture for WebRTC on a LAN with PSTN access ?   Olivier
• [asterisk-users] Experience with WebRTC on MacOS ?   Olivier
• [asterisk-users] On Register, run a script, validate source IP   Olivier
• [asterisk-users] Experiences with Jitsi's jigasi   Olivier
• [asterisk-users] Two interfaces, pjsip, 180 ringing contains wrong contact IP   Benoit Panizzon
• [asterisk-users] pre-dial handler, how to access variables from calling channel?   Benoit Panizzon
• [asterisk-users] On Register, run a script, validate source IP   Benoit Panizzon
• [asterisk-users] pre-dial handler, how to access variables from calling channel?   Benoit Panizzon
• [asterisk-users] On Register, run a script, validate source IP   Benoit Panizzon
• [asterisk-users] Check other calls on same endpoint (validate / screen customer supplied Diversion / From header)   Benoit Panizzon
• [asterisk-users] Global number rewriting rules affecting ALL headers?   Benoit Panizzon
• [asterisk-users] Global number rewriting rules affecting ALL headers?   Benoit Panizzon
• [asterisk-users] Global number rewriting rules affecting ALL headers?   Benoit Panizzon
• [asterisk-users] trust_id_inbound=yes but take CallerID(Num) from From: not from PAI   Benoit Panizzon
• [asterisk-users] On Register, run a script, validate source IP   Benoit Panizzon
• [asterisk-users] bug in pjsip trust_id_outpound?   Benoit Panizzon
• [asterisk-users] bug in pjsip trust_id_outpound?   Benoit Panizzon
• [asterisk-users] Avoid transcoding if both ends support same coded   Benoit Panizzon
• [asterisk-users] PJSIP device_state_busy_at, how does this work?   Benoit Panizzon
• [asterisk-users] PJSIP device_state_busy_at, how does this work?   Benoit Panizzon
• [asterisk-users] pjsip: How is asterisk choosing the IP address to put in the Contact header?   Benoit Panizzon
• [asterisk-users] pjsip: How is asterisk choosing the IP address to put in the Contact header?   Benoit Panizzon
• [asterisk-users] Centralized Voicemail Server   Alexander Perkins
• [asterisk-users] Dynamic Agents Unavailable   Alexander Perkins
• [asterisk-users] Dynamic Agents Unavailable   Alexander Perkins
• [asterisk-users] sip.conf host!=dynamic peer specific options (e.g. directmedia=off, transport=tcp) not working!?   Thomas Roos
• [asterisk-users] Antwort: Re: sip.conf host!=dynamic peer specific options (e.g. directmedia=off, transport=tcp) not working!?   Thomas Roos
• [asterisk-users] CDR extract call numbers on interval on unique callers   John Runyon
• [asterisk-users] Is it possible to record 2-4 party call audio in stereo quality as opposed to mono?   Antony Stone
• [asterisk-users] email notification on missed call   Antony Stone
• [asterisk-users] Truly *background* music on hold   Antony Stone
• [asterisk-users] Asterisk 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5 Now Available (Security)   Asterisk Development Team
• [asterisk-users] AST-2019-006: SIP request can change address of a SIP peer.   Asterisk Security Team
• [asterisk-users] AST-2019-007: AMI user could execute system commands.   Asterisk Security Team
• [asterisk-users] AST-2019-008: Re-invite with T.38 and malformed SDP causes crash.   Asterisk Security Team
• [asterisk-users] possible bug in Asterisk 16   Fourhundred Thecat
• [asterisk-users] possible bug in Asterisk 16   Fourhundred Thecat
• [asterisk-users] problem with logger   Fourhundred Thecat
• [asterisk-users] multiple softphone clients and same/different account credentials   Greg Troxel
• [asterisk-users] multiple softphone clients and same/different account credentials   Greg Troxel
• [asterisk-users] multiple softphone clients and same/different account credentials   Greg Troxel
• [asterisk-users] email notification on missed call   Frank Vanoni
• [asterisk-users] email notification on missed call   Frank Vanoni
• [asterisk-users] On Register, run a script, validate source IP   Jöran Vinzens
• [asterisk-users] Traffic police Cyberabad: Revoke 24 hours One way introduced at INDU RUB - reintroduce 2 way in the morning hours   kvsandeep445 at gmail.com

Last message date: Fri Nov 29 09:30:42 CST 2019
Archived on: Fri Nov 29 09:26:28 CST 2019

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