Archives : November-2019
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 ?..
I need some advice:I use 2 different suppliers of trunk SIP in my infrastructure, both send me regularly prices in a .csv format.So I have two SQL tables that contain the prefix and the tariff.For now, I generate a dialplan with a Perl script that all..
Hi. Does anyone have suggestions on how to achieve true background music on hold, by which I mean music which plays continuously, even in the presence of additional spoken announcements, such as: a) intermittent Playback() commands which I want to..
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 (..
Gang Next Problem which occurs. In Switzerland this is the common using form SIP Signaling: P-Asserted-Identity: Contains the provider provided and screened phone number which is the legal origin of the call. The origin which is to be billed for ..
One more Problem I stumbled upon. Using Asterisk in a TSP environement. Incomming IC Calls are e164 and have a NPRN (Routing Number) prefixed. Example: +4198055615995555 +41 country prefix 98055 Routing Prefix 615995555 effective phone number Calls rou..
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..
Gang Yes, big project on the rise to do things better / more flexible than our existing commercial TSP switch. During call screening process, we would like to allow customers to send the original callingID in a attended call diversion scenario. F..