Anyway, I guess that’s a minor bug if and only if the number I dialed is honored. Here’s the full config: # grep -v ^# /etc/jitsi/jigasi/sip-communicator.properties | grep -v then connected to, clicked on the + sign, typed in “4053”, and noticed that the jitsi-meet web UI keeps notifying me that " has been invited" (ie. Unfortunately, I was unable to see the above SIP extension 4901 with “status OK” with the default jigasi properties. Here’s my super simple Asterisk configuration: # cat /etc/asterisk/nfīoth SIP extensions seem to be OK: # asterisk -rx "sip show peers" I created two SIP extensions: 4901 (used by jigasi) and 4053 (a softphone I’d like to invite into a jitsi-meet conference room). That would be Asterisk 13 where I configured it as “basic” as possible, listening on port 5061 as port 5060 is already taken by Jitsi. So, just to make sure my production Asterisk server (1.4) isn’t the culprit, I decided to install the latest stable Asterisk version available on Debian 9 where I also have Jitsi-Meet & Jigasi. VERBOSE logger.c: Looking for s in from-sip-external (domain inf-voip) VERBOSE logger.c: - (12 headers 0 lines).
What I’m wondering is why is it trying to look up “s” instead of the number I typed when I was inviting someone from the web application (the plus sign). The “s” priority or extension does not exist in that context.
The problem I see is that Asterisk tries to look up s in the dialplan’s from-sip-external context. Inf-voip is the Asterisk server’s hostname. (from above example): // Call In Numbers and Codes It seems to need https for the json with the numbers.Īnd you need to specify both the API URL and the numbers url in your config.js, e.g. The way it worked for me is buried in here: Showing the dialin Numbers and the Conference number took a while for me to find out as well… in the middle, so you have a way of manipulating the headers (after asking the user for a conference number, for example).Īn example setup very similar to what I am trying is here: So, you are probably better of if you connect an asterisk or freeswitch etc. To my understanding (I struggle with jigasi currently as well) dial in directly from a SIP Provider will probably not work, as jigasi expects a custom SIP header in the INVITE Message of an incoming call, telling it which room to join.
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