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I am not familiar with version 5, but looking at this thread: https://www.pbxforums.com/threads/inbound-calls-dropping-instantly.6361/#post-26080 It would appear the name may now be "provider".

Mark updated this yesterday. https://github.com/fusionpbx/fusionpbx-install.sh/commit/7e62f45ff66731a83b4de723e9a37d2a3e0ac38f

When you make changes to the ACL make sure you "Reload ACL". A "Flush Cache" before the reload is also advised. Menu-> Status->SIP Status...

Not sure why the media attribute would be inactive, unless this is being copied over from the other leg, i.e from the phones invite. It is quite normal not to see this type of media attribute at all in an initial invite (inactive, sendonly, sendrecv etc.). Not sure why they are mentioning...

Hi Robi, No, the 401 Unauthorized is a normal part of the authentication process. In the example you show above, if you inspect the packet headers, you will see that the first REGISTER will not contain any credentials, so the response is 401 Unauthorized, but this 401 response will contain a...

I don't think it can be Fusion/Freeswitch related because the 603 Decline message comes from the phone. Maybe test it using different soft phones?

Hi Akashlal, The 200 OKs are in response to the OPTIONS messages, so you can probably ignore those. This is almost certainly an issue with the ring timeout on the soft phone. We had a similar thing with Yealink desk phones and solved it by setting: phone_setting.ringing_timeout = 600 Obviously...

Welcome, this is a good place to get help.

In your outbound route, you can set a SIP header using the syntax sip_h_<name> for example:

You will need to create another External SIP profile to listen on port 60, you will also need to change your firewall/fail2ban rules to allow connections on port 60. Also be mindful of the current QOS setting implemented in the firewall for the standard SIP ports.

OK, I see you are stuck with IP addresses. one option would be as you suggested to use one external SIP profile per customer. Obviously you will need to configure all of these IP addresses on your (~virtual) Network Interface Card. Avoid splitting addresses within the same subnet across...

This seems very wasteful of IP addresses, sure it could be done, but why not just use registration in your gateways. Registration will still keep the contracts between carrier and tenant.

If you have a reasonably good internet connection then you should have no trouble with a call from desk to desk going via the internet. We have sites with 300+ phones all talking to each other via the internet. Make sure your internet connection is good, you have a decent reliable router, and...

I don't know much about multicast, but I'm pretty sure you can't do it over the public internet. Multicast routing is different and all the routers in a given path need to know how to handle it in order for it to work. With UDP, you can do a pseudo multicast by simply firing the same packet to...

You can call another process such as a script form the dialplan. Have a look at /usr/share/freeswitch/script/cidlookup.lua This is an .lua script that can be called on an inbound route to lookup a callers name from the contacts table. I sure this could be adapted to interface with CRM.

Hi Lukas, In your screenshot the ACL name is "providers", this may be a Fusion V5 thing that I am not familiar with. What is the name of the ACL specified by apply-inbound-acl in your external profile?

It Looks like your PBX is asking the SIP trunk provider to authenticate when it receives the invite from the gateway. This does suggest an ACL issue. Also check in your external SIP profile that auth-calls is set to false. (No sure what the new defaults are for FP v 5).

It looks like a network related issue, just like the type of thing experienced with NAT problems. Try to find out why the INVITE is sent from port 1024 but port 1025 is advertised in the Contact header.

Ok, so in the dialplan manager you should be able to set a catch all, the regex would be something like ^[0-9]+$ (see the attached image). Make sure you set the dialplan order to something like 999 so it will not interfere with other dialplan rules.

You shouldn't need to do anything, by default the system will respond with "Temporarily Unavailable".