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It probably depends on the phone. You could try putting the IP address in as a proxy server. Personally, I would stick to using DNS, its not troublesome. You could use a decent router like a MikroTik to provide DNS or set up a caching DNS server (bind) on your PBX server.

@hfoster Good you should mention eavesdrop. The .lua below was written to emulate an old fashioned answering machine using voicemail and a standard extension, where you can listen to the message being left and pick up if you want to. This may serve as an eavesdrop example for HeisenbergQin...

My customers use this facility on the Yealinks a lot.

I think in situations like this, the more options we have to explore the better. It helps to generate ideas for better ways of doing things.

Just interested, what alternative would you recommend for a low bandwidth codec? (I suppose GSM is the other obvious one...)

You could certainly do what you require with an .lua script. I'm pretty busy at the moment but if I get time I will dig out some examples for you.

G729 is now royalty free, you just need the shared library mod_bcg729.so installing in to your /usr/lib/freeswitch/mod directory. This thread may be helpful: https://www.pbxforums.com/threads/install-mod_bcg729.247/

I would expect to see invites from multiple ones, but in order of preference. So when an inbound call comes it it is routed via preference 1, if that INVITE fails for some reason, it is routed via preference 2, and so on... Sending simultaneous (multiple) INVITES in this scenario is a waste of...

I think your problem is having two registrations with the provider and the provider setting them both with the same preference weight. I would normally only expect to see one registration with any given preference weight.

Conference bridge?

We may be able to do something, not quite sure, will need to check with our ops people. Is it outbound only or termination as well, also what will be the anticipated call volume? PM me if you don't want to share the information publicly.

Are you looking for something at a wholesale level or just a SIP trunk terminated in the UK?

It sounds like a problem with your dialplan, check what matches 6xxx, something is picking it up and asyou say probably stripping the first digit. Also carefully check anything that is global or runs in the public context i.e. inbound routes.

It's worth looking at call limit in the dialplan manager you can do a lot with it: https://docs.fusionpbx.com/en/latest/dialplan/dialplan_manager.html Also look at the documentation for the actual limit function in mod_dptools on the FreeSWITCH Wiki...

Welcome.

You are very welcome.

Yes, I often recommend packet captures when diagnosing SIP, so much so that I begin to sound like a broken record! (Reference for those who remember scratches on vinyl that made the needle jump). ngrep, sngrep, tcpdump and Wireshark are the tools to help you. Look closely at the Contact...

Welcome. You well get plenty of help here. Look out for @ad5ou he is relatively "Local" to you and is very knowledgeable with Fusion and FreeSWITCH.

It is unlikely to be an issue with the phones unless any configuration has been written to any of the static fields. You can check this by exporting all the configuration from the phone(s) and looking at it with a text editor. Look out for items that would not get overwritten by the provision...

There are probably many different ways you could achieve this. A .LUA script to query the FreeSWITCH Sqlite DB would probably be the most flexible for you. You can test this at the command line with: root@testbox1:# /usr/bin/sqlite3 /var/lib/freeswitch/db/core.db "SELECT COUNT(reg_user) from...