Search results

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.

Its basically just experience. If you read through it, what you can actually see is that it is going through the dialplan as defined in the dialplan manager.

No, your outbound routes are likely to break everything, you are saying send anything over three digits, that will likely include ring groups, extensions, etc,etc, etc, Create separate outbound routes to match your emergency numbers etc.

I believe you problem is here: bridge(sofia/gateway/5719458e-8222-4631-b62e-d7c64d5fef69/7000 Why is it trying to use a gateway to send that call? I would guess that you have an outbound route that is catching that.

Paste a freeswitch log of one of these calls

In the advanced gateway settings try setting callerid type to pid, again, after the change, stop and restart the gateway. Also, can the provider not help you out with this? They should be able to tell you what the invite should looks like.

In your gateways advanced settings, do you have "Caller ID In From" set to true. If not can you set that then stop and restart the gateway.

Hi, I see the problem immediately: 2024-09-29 23:20:59.357307 98.63% [DEBUG] sofia.c:10582 IP 108.60.224.235 Approved by acl "providers[]". Access Granted. You have modified the ACL that should not have been done, the PBX is now treating you as a provider and it is using the public dialplan...