Configuration of a self hosted jitsi jigasi for incoming calls

Hi there,

I am a cofounder of a startup in France, and we needed an open source self-hosted videoconference service. Jitsi sounded perfect. Our IT service provider installed it on a Debian and it works like a charm.

We are now polishing the configuration, and we would like the conference room the be reached by incoming calls, ideally someone just dials a phone number, another number for the conference room, and voilà. We have an SIP trunk, the service provider configured it. Outgoing calls do work. I am now trying to figure out what it takes to be have a conference room that can be reached by an incoming call, right now the call doesn’t reach the room. It is unclear to me what is the way to go.
Option 1 seems to be use a conference mapper (see Conference Mapper | Hosted Jitsi and [jitsi-users] Telephone dial in support) but I fail to find an updated documentation on how to do it…
Option 2 could be to manually change the SIP header in the INVITE Message of an incoming call, telling it which room to join. That seems quite painful and not very efficient…

Last but not least, I searched for a way to support you guys financially but I could not find it. How can I do that please?

Thank you very much for the great work! Please let me know if I can provide any more information.


This is the option. You use the conference mapper to map meetings to numbers( IDs ) which then people dialing in can input in an IVR and based on that number you can instruct jigasi through sip header in which room you want it to join.

Damian, thanks. I am far from being able to code that myself, is there an existing implementation out there? I assume there is quite a bit of code to write, no?

For conference mapper you can use the one that is used by
For the rest, IVR and stuff there were few examples in the forum using asterisk … it all depends on your tools and providers, so I cannot give you an advise.

ok I will ask them. Thanks.

So it turns out the SIP configuration is not trivial at all, and a basic SIP trunk cannot achieve this. I have opened this request for better documentation:

What is the problem you see?
It is hard to improve if you don’t know what?
The current installer creates a general sip config that only may need touching the codecs and that should work. Mind that jigasi works as user agent, as a sip phone. It cannot work with sip trunks.

What is the problem you see?

here is the deal: (in France at least) when you ask for an SIP number, SIP providers provide you with an SIP trunk and a phone number. We have that. And btw it does work when the room specified under ‘org.jitsi.jigasi.DEFAULT_JVB_ROOM_NAME’ is configured properly as mentionned here. but this is valid for one conference room only.

I had my SIP provider and IT service provider talk together, the SIP provider told me that having a tool where a guy dials a phone number configured in Jigasi and then the pin of the conference room to join that specific conference room won’t work. He told me it was possible using some proprietary softwares, but nothing either directly available or easy to work with from scratch. He is absolutely aware of the ened to customize the header as described here. The IT service provider did do his part of the job, he used the conference mapper that is used by, I can copy paste here what he did if that is useful.

I have 3 options:
1- I totally give up and use whatever proprietary not self hosted service
2- I keep the jitsi instance and give up on the phone number. This is not going to work on the long term
3- I keep on digging, and for that I need a better documentation. I might have missed something, but on the jigasi page I failed to see any specificity for the SIP client. I think this would quite useful. What is obvious for people who have worked long term on a project may be not that obvious for others.

I have a quite light background in IT, I am just trying my best to make sure my company operates with open-source and not GAFAM hosted softwares.

Hello @edouard.duliege

Could you please share your current configuration? I’m already happy when we can dail-in one room.

Thank you.

Hi Syan, this is quite straightforward, look here. If you still have a problem, please open another thread.