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Thankyou Daz somtimes i should trust myself. Do you think its our IP its rejecting. Is technically the 403 the reply from the carrier Kind Regards Andy

@DigitalDaz Invite 100 TRYING 403 ACK

{sip_invite_params=user=phone} Hi Digi Just been test with the providor as although we are now sending the right invite its not even hitting them with this in the bridge statement. As soon as i remove the call can be sent again but ofcoarse it fails as they want to see the correct invite...

Thankyou that nailed it! Still gto a forbiddon respnce from them but its now exactly how they have asked for it. Thankyou and Matt for all your help!

Sorry i mean URI they want to see user=phone after SIP/2.0

Ah thankyou I'll test that. Any ideas how to add the tag in the url header user=phone Kind Regards Andy

In the FROM line in an invite for a call. Ive attached a shot of the beginning of the line.

Hi Digi Thanks for the steer yesterday. Im on outbound now and managed to adjust the invite to the providors requirements except a couple of things 1. They dont want to see the Ext at the beginning of the FROM 2. They would like to see user=phone in the URI header. Any advice - Matts also...

Can you see this registered properly in - STATUS - REGISTRATIONS Is account 999 showing in there online?

Place you ddi in Dialplan =) destinations and use dropdown to select fusion destination (Ext will be in here)

Ahhh what a newbie mistake to make. Sorry digi

@DigitalDaz Ok got an inbound example. Calls coming through now to the platform and i can answer but i cant hear any voice. Here is the inbound invite - is there any chance you could take a look. Happy to take this up direct with you officially.

Ive had this put in for a client before :- https://okay.network/communications/lcr-for-fusionpbx-with-freeswitch.html

Ive sent them the trace for what we can see as to be fair it looks fine on first look. Mean while they also have specific requirements for making a call and im struggling getting the FROM format correct. It needs to be - From: "6666" <sip:FreeSWITCH@46.***.***.106>;tag=pZ6yXv8BaS8Zc...

I havent touched the ACL but ive just asked them to send to 5080 and they are now getting a 480 temp unavailable reply

This ones a little different. We are working with a new to Sip providor. They are sending us a test invite but i cant even see it hit our side this is what they have sent us but im reaching out for wisdom please. Is there invite correct for us to pickup the call. Ive **** ome of the details...

Hi - More details please Where are you hosting this? What server OS are you building? If server is going down after you login i would suggest a different host. Installed hundreds of Fusion instances in the cloud and never come accross this. Can you run a ping while you login? Is the server...

I would upgrade your fusion as a starting point. Its on 4.4.2 now and there were a load of bug fixes / security fixes from 4.4.0. Retest and then ill attempt to simulate should you have similar issues. Regards Andy

We has this put in Version 4.2 *** Not tested in latest version **** This created a copy of the call recordings, renamed them to the logic you descibed and saves them in seperate area. This way the original was also still in fusion and could be searched via the UUID. At the time it was a big...