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OK, if you are entering the correct provision username and password, then we may need to gig a little deeper. The 401 is an unauthorised error for sure. You can set the provision username and password either globally in default settings or an the per domain basis in domain settings. Make sure...

Sorry, I'll re-phrase. What ever username and password to put into the http_auth_username and http_auth_password variables. All you need to do is type the same into the sign in dialog when it is presented to you. You will then be authorised to see the config files that your phone would download.

The provisioning is asking you to provide WWW-Basic-Auth, which is the method used by the phones. It is different from the auth. mechanism used in the Web GUI. All you need to do is type in the http_auth_username and http_auth_password that you have used in the default or domain settings.

Very basically, if an endpoint registers using UDP, FreeSWITCH will report the registered status as UDP. If, however, FreeSWITCH, or more specifically mod_sofia detects any RFC1918 addresses in specific headers, Contact: for example, it will report the registered status as UDP-NAT. FreeSWITCH...

If I remember correctly the mobile-twinning app was written by @KonradSC. If you search on here for mobile-twinning you will find a few posts about setting it up, mostly from around 2021. I'm not sure if it is maintained, but we certainly got it working when we had a play with it on Fusion 4.5...

So if I read your sngrep correctly, your PBX makes no attempt to send an INVITE to your provider for the external number in the ring group, yet your application logs show a bridge to a mobile number via your gateway. I'm wondering if your sngrep is missing some information, especially as I...

What am I looking at? Are these successful calls? They don't look like it. Is this between your PBX and provider?

OK, it may need the name or IP of the Proxy in your case. The best way to check is to run sngrep and make a call out on that gateway that you know works ok. Then you will see what is being used as the invite domain for a successful call.

Hi, this is just a guess... Try setting sip_invite_domain in your outbound route. Set it to whatever you have your "from domain" set to in the gateway record.

You may need to look at a dialplan record called bind_digit_action. https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Modules/mod-dptools/6586414#0-about

@mcs3ss2 Of course RDNS - sometimes we forget to state the very obvious! I get all the DMARC reports but I didn't know Microsoft 365 actually checked for it. We learn something new every day...

You may need to look at email headers in detail, and compare the working ones with the non working ones. As I said, some providers are doing non industry standard things like silently dropping emails (No non delivery response/reason) that they don't like to look of. For example, I have seen...

The big providers are making it increasingly difficult for people running their own email servers. Some are not following IETF guidelines and are ignoring the RFCs on the basis that "they know better". Just make sure that when you send emails from an authorised account on your mail server...

@seankann Yes, you get the idea, probably no need to sleep after every endpoint, maybe every 10 or every 100, see code snippet below. And as I said you only really need to worry about this if there are a large number of endpoints. import time i = 1 while True: i += 1 if i > 10...

This is where all GUI wraps can cause confusion, not just the PBX you are using. An outbound route is really just a dialplan entry like any other. I don't use Fusion now but there used to be a menu item called Dailplan->Outbound Routes Here you used to be able to add as many extra lines as...

Agree with voipBull, but will just add, if you don't want to create another outbound route you can just bounce through the dialplan without creating an actual extension, see image below, this creates a dummy extension 6843 and also allows you to set channel variables the same as you may do in an...

Be careful running this kind of script on a big system, you may want to consider dividing groups of endpoints into chunks and put in a small delay between each chunk for two reasons: 1. You don't want to create an sustained outbound SIP message load that may saturate mod_sofia. 2. If you run a...

https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Troubleshooting-Debugging/Hangup-Cause-Code-Table_3964945/

It will largely depend on: A. What else you have set in the inbound route/dialplan. or B. What the calling system is programmed to do with a USER_BUSY response.

There are (were) some Linux deleted file recovery tools, I used them many many years ago with some success but can't remember the details now. It will probably depend on your filing system type, and it is also very important that the disk is not written to until you recover the files. I would...