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Another possibility is that you could have some sort of ALG operation on your router that detects the flow of packets to port 5060. Probably worth checking your router configuration. DigitalDaz makes a good point about iptables and fail2ban, as you are getting successful registrations it is...

I would argue that it is not good practice. I guess it would be fine and not too difficult to manage if you only had a very small number of gateways. I would ask why the SIP provider needs to see different ports, there really should be no need. If your gateways connect by registration, the...

The SIP port used by your gateway is the port specified in the external SIP profile. The only way you could have different ports per gateway (without introducing a proxy like OpenSIPs etc.) would be to set up multiple external SIP profiles.

If registration and SIP signalling (i.e call set-up) is working then your SIP port configuration is probably fine. Audio/video is sent using the RTP protocol, not SIP. Whilst you do specify what port range may be used for RTP, the actual port is determined on a per call basis and communicated...

I'm glad you managed to sort it yourself, please post what you did, it will help others.

Just check what you have in terms of permissions in the extension category for the superuser. Because the master branch is a moving target, I can't say what all the permission may be at this moment in time, but you may want to uncheck "outbound_caller_id_select" and make sure that...

Your regex for the destination_number is not correct if you are trying to dial 0015939... Do you prefix international calls with 00 or 001? Assuming it is 00 and you wanted to call the UK you would dial 0044... In which case your regex could look something like this: ^(00\d{9,17})$

For the internal profile, ext_sip_ip, ext_rtp_ip should be set to whatever IP address you expect your phones will see packets coming from. Normally the defaults are fine. The situation that I addressed for Albert-g was more complicated than usual because three IP address ranges, internal...

Many of us do have the a master branch version in production, not the very latest, but a version, that, as an individual, we trust through our own testing. I don't think anyone is comfortable with the current situation, but FusionPBX remains a great product and we keep contributing to it...

We have forgotten what a stable version looks like. The whole world and his dog are eagerly awaiting V4.6, which we hope will be the next stable version.

I can't really help, but bear in mind that it's not all about CPU usage, you have to consider other factors like I/O blocking. You don't say what codec is being used and if it is the same on both legs of the call (transcoding). The type of codec does have an impact on the amount of work the...

Difficult to diagnose further without a proper packet capture, screenshots of sngrep are of limited use. You may have a NAT detection /Traversal issue. Why does the INVITE go to 41@1 something when 41 is actually 41@v something? Are 20,22,21,1,10, and 23 all on the same public IP address? If...

What it is actually called can very from router to router depending on the manufactures terminology. Generally is is referred to as "NAT Session Timeout Settings". How one sets it also varies tremendously from router to router, for example on a Draytek it can not be set in the GUI, it must be...

We could write a whole book about securing a hosted PBX, so it is very difficult to answer your question with a simple forum post, here are a few pointers... By default FusionPBX protects itself quite well, but improvements can be made, this is true of any system and often what you actually do...

Hi Albert, So we can see that for this packet MTU and fragmentation should not be the issue because we can see the Content-Length header is 549, also and dependent on where the packet was captured (in your case the server so not relavent for this diagnostic) we can also see the closing...

Much of that detail is stored within FreeSWITCH at runtime. You can query what is stored at the fs_cli: freeswitch@myserver> find_user_xml * 201 me.mydomain.uk If the details are wrong, try a flush cache and reload XML.

The first thing I would do to diagnose this would be to get a packet capture both on the network the PBX server is connected to and the network the phone is connected to (you can do small captures on Yealink phones form the Settings->configuration page). My guess would be that the notification...

In the UK, there are several companies that supply empty 19 inch rack mount enclosures, here is one example below: https://www.metcase.co.uk/en/Enclosures/19-Enclosures.htm I'm sure you will have similar suppliers in Italy.

Ok, it is fairly straight forward for you then, just set the sip-ip, rtp-ip and ext-sip-ip and ext-rtp-ip directly to whatever you need in the SIP Profiles: Note: the variables that look similar to $${local_ip_v4} are just FreeSWITCH variables, in this case the local IPv4 address that it has...

Yes, of course you can. Both internal and external profiles can use the same public IP as long as they are on different ports, they are by default (5060 and 5080). If your FusionPBX is going to be behind a NAT, then you will have a little more work to do, but there is plenty of documentation...