Messages by Thread
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[asterisk-users] Asterisk 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5 Now Available (Security)
Asterisk Development Team
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[asterisk-users] Experiences with Jitsi's jigasi
Olivier
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[asterisk-users] Low cost routing
Olivier CALVANO
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[asterisk-users] Truly *background* music on hold
Antony Stone
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[asterisk-users] Experience with WebRTC on MacOS ?
Olivier
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[asterisk-users] Which architecture for WebRTC on a LAN with PSTN access ?
Olivier
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[asterisk-users] trust_id_inbound=yes but take CallerID(Num) from From: not from PAI
Benoit Panizzon
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[asterisk-users] Global number rewriting rules affecting ALL headers?
Benoit Panizzon
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[asterisk-users] WebRTC: which ACL and modes for /etc/asterisk/keys/asterisk.pem ?
Olivier
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Re: [asterisk-users] Question on WebRTC configuration
Olivier
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[asterisk-users] How to set http.conf for HTTPS support on Debian Buster ?
Olivier
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[asterisk-users] Check other calls on same endpoint (validate / screen customer supplied Diversion / From header)
Benoit Panizzon
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[asterisk-users] On Register, run a script, validate source IP
Benoit Panizzon
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Re: [asterisk-users] On Register, run a script, validate source IP
Sebastian Nielsen
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Re: [asterisk-users] On Register, run a script, validate source IP
Benoit Panizzon
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Re: [asterisk-users] On Register, run a script, validate source IP
Olivier
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Re: [asterisk-users] On Register, run a script, validate source IP
Jöran Vinzens
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Re: [asterisk-users] On Register, run a script, validate source IP
Benoit Panizzon
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[asterisk-users] Does Asterisk support one-legged transfers with external switches?
Dan Cropp
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[asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer
Sebastian Nielsen
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[asterisk-users] Asterisk 16.6.1: PJSIP: delayed action of core since update to 16.6.1
O. Hartmann
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[asterisk-users] problem with logger
Fourhundred Thecat
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[asterisk-users] What is Debian Buster's asterisk equivalent of Digium's contrib/scripts/ast_tls_cert ?
Olivier
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[asterisk-users] pre-dial handler, how to access variables from calling channel?
Benoit Panizzon
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[asterisk-users] Centralized Voicemail Server
Alexander Perkins
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[asterisk-users] SIP messaging
Jerry Geis
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[asterisk-users] Digium's Opus Codec download links broken?
Floimair Florian
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[asterisk-users] Two interfaces, pjsip, 180 ringing contains wrong contact IP
Benoit Panizzon
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[asterisk-users] INFO to rfc2833 issues
Dovid Bender
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[asterisk-users] CDR extract call numbers on interval on unique callers
Andre Gronwald
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[asterisk-users] sip.conf host!=dynamic peer specific options (e.g. directmedia=off, transport=tcp) not working!?
Thomas Roos
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[asterisk-users] Traffic police Cyberabad: Revoke 24 hours One way introduced at INDU RUB - reintroduce 2 way in the morning hours
kvsandeep445
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[asterisk-users] possible bug in Asterisk 16
Fourhundred Thecat
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[asterisk-users] Is it possible to record 2-4 party call audio in stereo quality as opposed to mono?
Dan Cropp
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[asterisk-users] Own MOH incorrectly kicking in instead of the MOH of the callee
Michael Maier
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[asterisk-users] Stuck "channel"
Carlos Chavez
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[asterisk-users] No audio after hold
arish haque
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[asterisk-users] email notification on missed call
Fourhundred Thecat
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Re: [asterisk-users] Multiple readfile oddities, newlines etc
Jonathan H
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[asterisk-users] Fwd: [CFP] FOSDEM 2020, RTC devroom, speakers, volunteers neeeded
fosdem-rtc-admin
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[asterisk-users] Asterisk 17.0.0 Now Available
Asterisk Development Team
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Re: [asterisk-users] defaultexpiry & maxexpiry on peer level
Jonas Kellens
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Re: [asterisk-users] AGI GET DATA same DTMF code as Read cmd ?
Joshua C. Colp
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[asterisk-users] issues with Asterisk CLI
Fourhundred Thecat
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[asterisk-users] ConfBridge and sound prompts
Dan Cropp
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[asterisk-users] Realtime PJSIP max_streams' issues
Ahmed Chohan