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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...

Just one line should do: action limit hash myinout ${domain_uuid} ${max_calls} !USER_BUSY

Your fist one should work but you will need to set the "inline" flag on the set action otherwise the max_calls variable will not be available for the limit action. You may also want to use a more simple name for the realm without the (, |, and) characters.

Have a look at https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Modules/mod-dptools/3375201/ When you say "limit the number of incoming and outgoing channels per domain", I assume you mean inbound and outbound calls via gateways not simply channels connecting two extensions. If...

Unless it's changed in newer versions of FusionPBX, you can just select any of the ${sip_ choices and then click on it to edit it to whatever you want. Screen shot below:

Yes, so the lower capture confirms it, there is fragmentation going on. The choices are 1. Switch to TCP (bigger system overhead) or 2. Reduce your UDP packet size. One of the easiest way of reducing the UDP packet size is to limit the number of codecs you offer, thus reducing the size of the...

Maybe start by looking at your UDP packet sizes. If the packet size is exceeding the MTU, then it will be fragmented, TCP is unaffected by this. I have seen it many times when an initial INVITE message is OK, but then the proxy responds with w WWW-Authenticate, the client sends the INVITE...

@alan If I understand your original question correctly, you are asking how to set the caller_id in the dialplan, if so, then just set effective_caller_id_number=+1xxxx etc. If you are trying to make a call from India, look like it originated in the USA, your call may get blocked or have the CLI...

I apologise for butting in here, I noticed that in your original post you mention cross talk, it was common in the analogue telephony days, but now few people take reports of it seriously with VoIP, however there are ways it can happen. If nothing had changed in your Fusion/Freeswitch...