Sipxcom and Jitsi video conferencing with jigasi

[Arturo J Garcia Jmz]

Do some one have any experience integrating sipxcom with jigasi in a jitsi server?

So far now I was abe to sign in the jigasi to receive calls to the default "siptest" room, but I can not do any call.

In other way I´m looking for information to have better undertanding of the URI´s, so I can setup and IVR for this conference room and set up the dial plan configuration I need to acomplish this .

[maxg...@gmail.com]

Hi,

yes, i got it fully working with a lot of looking into the code and customizing. Dialing into a Jitsi conference and dialing out of a jitsi conference (invite a phone number) is working.

Jigasi registeres itself as "normal user" at sipxcom, so you have to create a user for jigasi and provide username, password, domain, etc. in the jigasi config.

To dial into a specific jitsi room (other than default siptest) you have to set an additional sip header in the SIP INVITE Message to Jigasi. The Header is: "Jitsi-Conference-Room=<jitsiRoomName>". Dialing into a empty Jitsi room is not possible yet due to a bug in jigasi v1.1. in case of an empty room, jigasi doesnt answer the call.

These are just approaches. There is many more magic behind the scenes, which is to complex to explain here. 

In conculsion: jigasi isn't ready yet for working out of the box. You have to customize a lot on jigasi and sipx site.

Best

Max

[Arturo J Garcia Jmz]

Thanks Max

Do you have something you can share with me, I have setup ready to be customized, but I do not know how start to do that on the "INVITE"  or what to check to make call from the jitsi conference room

The reception of call to a conference room it will be my goal!

Thanks

[Matt Keys]

If you're looking for audio conference the default freeswitch conference works well. There is a "Enable Video" option in the conference room settings you might try out. Of course there's always eZuce Vibe. :) Outside ezuce/sipx stuff I've had success with mediasoup ( https://mediasoup.org/ ). Phantom users with 'if no response' call forwards are perfect for routing INVITEs into stuff like that. The call forward target can be a sip uri like confe...@foo.org, but I don't think you're allowed to specify parameters like confe...@foo.org;Jitsi-Conference-Room=foo. If you need to specify special parameters you might need something like a lua script (features - ivr, specify type lua, documentation https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference ), or forward to something that then forwards again with the parameters you need.

[Arturo J Garcia Jmz]

Thanks Max

It is the Lua freeswith already part of sipxcom?, I know the freeswitch is, but the lua is ?, how do I acces it ?

[Todd Hodgen]

Under Server services, enable conferencing.  You will find the option once enabled under system/features.

sent using my two left twiddling thumbs

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From: sipxco...@googlegroups.com <sipxco...@googlegroups.com> on behalf of Arturo J Garcia Jmz <arturojg...@gmail.com>
Sent: Friday, May 15, 2020 10:45:12 PM
To: sipxcom-users <sipxco...@googlegroups.com>
Subject: [sipxcom-users] Re: Sipxcom and Jitsi video conferencing with jigasi

 

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[Matt Keys]

My bad, I guess that's exclusive to Uniteme now. It seems like that used to be in sipxcom around 14.04 or 14.10 under call queue settings. I see JIRAs related to it on the sipxcom side in 16.08 (sipx-507 http://wiki.sipxcom.org/display/sipXcom/sipXcom+16.08).

[ابو علي‎]

What are the available ports?

[Matt Keys]

Check beneath system - security - firewall beneath the rules tab. Each port used by sipx is listed in the rules with a description. Freeswitch default port is 15060.