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I'm not sure if anyone will be able to help here, I for one don't have a cluse how fusionpbx integrates with dsip router. Its probably a question for dsip router people.

Thanks, that appears to have done the trick, modules now works, I assumed I needed to do a git pull first:

Yep, looks like a few more are missing:

Adding the correct path fixed it.

That was a new install yesterday by the way.

Yep, it is blank, I should have figured that out myself. I'm getting rusty :)

@pbxgeek I also found the modules page blank on a new install. It would appear that in resources/classes/modules.php, $this->dir is empty, populating it with '/usr/lib/freeswitch/mod' resolved the issue. I don't know where it pulls that in from, hope that helps a bit.

OK, still a bit scabby but create a file: /etc/fail2ban/jail.d/ignoreip.conf NOTE: you need each ip address after the first line indented with a space. Inside it, this kind of thing: [DEFAULT] ignoreip = 127.0.0.1/8 215.155.52.118 179.22.139.77 182.63.140.77 182.63.142.77 183.63.141.77...

I know I'm on of the people who pushed for the reinstatement of the wiki at that time, I'm actually suprised that it is still up now. That was long before the documentation was as extensive as it now is. At the time, there was a lot of very relevant information though I wouldn't dream of using...

Just try 880241100300 as the username

Awesome Adrian, its come a long way. Just a quick question: is the rabbitmq a single point of failure? I remember this being an issue with kazoo back in the early days.

What relevance is this to you? 0005975999 That appears to be the number it is trying to match but I see no reference to that in the invite.

Start by using sngrep to see what the INVITE looks like that the carrier sends you and post that.

I use a heavily customized version of pyfreebilling for years, that does LCR too.

@markjcrane Thanks for pointing this out, I had completely missed it.

So create two routes for the same dialplan with different gateways. Then for each of those routes, add a condition at the beginning that matches the extensions you want. The first condition will be something along the lines of sip_from_user ^200$ || ^201$

You do not appear to have outbound routes.

Well, immediately I see some of the filters are not enabled: [sip-auth-challenge] enabled = false <-------------------------------- port = 5060:5091 protocol = all filter = sip-auth-challenge logpath = /var/log/freeswitch/freeswitch.log #logpath = /usr/local/freeswitch/log/freeswitch.log...

CAn you provide a siptrace of an outbound INVITE from sngrep

Well to be blunt then what is ANI? Where is this set? According to the SIP specs, caller id is either set in in the FROM header or it is set in RPID, p-asserted-id etc. Some proveiders will use the from, some rpid, some pid (p-asserted-id) I know of no field in SIP for ANI, please enlighten me.