Ive just discovered jigasi : a server-side application acting as a gateway to Jitsi Meet conferences. Currently allows regular SIP clients to join meetings and provides transcription capabilitiesHave someone used it with Asterisk ?How does it work ?..
Author : Olivier
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Reading [1], I would be very curious to read about WebRTC on MacOS, either for Voice or Voice and Video calls.How does MacOS compare today to Windows or Linux regarding WebRTC support ?Do you need to use Chrome or Firefox to get WebRTC ?[1] https://webkit.org/blog/8672/on-the-road-to-webrtc-1-0-including-..
I would like to offer end users in a LAN, asking for this (why ? I dont know) the capability to use a laptop (along or in replacement of hardphones) to emit and receive PSTN calls.PSTN pass through a plain SIP trunk which does not support video (..
Following [1], you get precious help for webRTC installation.Something that is missing there, though, is a note expliciting/etc/asterisk/keys files ownerships and modes.As people are either running asterisk as root:root, asterisk:root and others or..
Ive installed a new Asterisk 17.0.0 on a Debian Buster system.This Asterisk instance is run by asterisk user (and group). Ive got:# ls -l /etc/asterisk total 68-rw-r–r– 1 asterisk asterisk501 nov.18 19:12 asterisk.conf-rw-r–r– 1 asterisk asterisk..
With Debian Busters asterisk package, what can you use instead of Digiums contrib/scripts/ast_tls_cert ?If that matters, this is for using WebRTC and Cyber Mega Phone 2K (both on the same box) in a private LAN environment.My intent was to use easy-..
Ive got an Asterisk 11.13.1 system running on a Debian Jessie platform. This systems extensions.conf doesnt include any reference to PJSIP, yet(only using chan_sip at the moment).This morning, it failed with:Aug 26 09:07:33 foobar kernel: [6534231.7764..
Ive just read in [1] about SIP MESSAGE addition to both chan_pjsip and ConfBridge. It seems very interesting addition as it brings the capability to mix voice, video and text in conferencing.On an other hand, there are some softphones (Zoiper, Br..
Is it possible to find real domain names instead of IP addresses in SIP URI?For instance, in a book dedicated to SIP (Understanding the Session Initiation Protocol), Im reading an example of a SIP INVITE that looks like:INVITE sip:411@salzburg.at;user=ph..
These questions crossed my mind this morning :In general, are anonymous international calls allowed (ie calling from one country to a number in an other country while hiding your own caller id) ?Are there special rules in Europe for this ?Be..