Hello Jacob,

I have installed Jitsi Meet with nginx on Debian 10, works fine. I want to install Jigasi so people can call into the meeting. I was going to follow the instructions at https://jitsi.github.io/handbook/docs/devops-guide/devops-guide-quickstart#adding-sip-gateway-to-jitsi-meet, Adding sip-gateway to Jitsi Meet.  It says during installation ofJigasi I will be prompted for a SIP account and a password. For a "free as in freedom" software person can you recommend a SIP provider and software to use?

Is there any way to make it  possible for someone to call in using a regular PSTN phone number or does that take a special interconnection layer?

Thanks,

Dennis

Hello Dennis,

The only commercial SIP provider I've used is SIPStation, which is part of Sangoma, the company who makes FreePBX. Supporting SIPStation would be supporting free software in that regard. You can find SIPStation here: https://www.sipstation.com/

The only bad thing I've heard about SIPStation is that its pricing (which is basically $25/month per unlimited trunk) can be less optimal if you don't use your trunks often. It may be more economical to use a pay-per-minute SIP provider. The company I set a FreePBX server up for was coming from a traditional phone system and wanted a flat rate, so this wasn't an issue for me, but it's something to consider.

I'm not sure what you mean by "special interconnection layer" for a "regualr PSTN phone number." The SIP provider provides the connection between the telephone network and your digital network. It's called DID: https://en.wikipedia.org/wiki/Direct_inward_dial

If you're thinking you're going to "game the system" by using a regular phone line instead of a SIP provider, you would need some sort of appliance to convert the analog line into a SIP line, basically being your own SIP provider. There are some of those devices on the market, although the ones I've seen are more to provide an emergency fallback in case your internet connection goes down and you still need to have your SIP devices work:

https://www.sangoma.com/voip-gateways/analog/
http://www.grandstream.com/products/gateways-and-atas/voip-gateways
https://www.voipsupply.com/voip-gateways/analog

When setting up Jigasi for your VoIP network, properly configuring spam URL removed by moderator (DID) is crucial to enable external callers to connect directly.

Jigasi will act as your SIP trunk handling call routing. First set it up with your VoIP provider for reliable connectivity. Confirm you have DID numbers allocated to use which should also be configured under your account.

In the Jigasi configuration file, ensure the "enabled" property under "sip" is set to true. Also provide your username, domain, password for your SIP trunk account from the provider.

Additionally, map each assigned DID number to your extensions using the "numberToExtensions" mapping table parameter. This links a specific incoming number to ring an internal extension.

With DID routing established in Jigasi mapped to extensions, callers from outside your network dialing your DID numbers will connect straight through to associated devices. Jitsi will consume these configured mappings to enable direct dialing.

Test with your team members first to validate inbound call handling through the SIP trunk. DID configuration is key for enabling this seamless external calling functionality.

Let me know if any issues crop up while testing! I’m happy to assist further with your Jigasi setup and DID implementation details.