Sip Trunk with outbound proxy

Hi guys
I ran into a problem with the sip trunk in issabel, which there is not much information about on the internet, but it seems to be a very simple solution. The sip phone settings that I received are a little different from the usual sip phone settings. In the sip phone settings delivered to me, there is a parameter called outbound proxy, and there is @ and domain in the username, and I think these parameters caused sip trunk not to run. I ran different softwares to test so that I could hear the beep of this sip phone, the only software that I could test the line correctly with all my effort was 3CX software. I will send the settings file and photos of 3cx settings here I have also managed to do trunk on Huawei FXS. If anyone knows how to run this sip trunk in issabel, please let me know. I would be grateful if you could advise me how to fill the PEER Details, USER Details and Register String sections.

Thanks before.

Image of 3CX Config and Huawei FXS:

3Cx Config :
Name=+98513******
CallerID=+98513*****
AuthUser=+98513*****
AuthID=+98513*****@r-khorasan.tci.ir
AuthPass=******
PBXRemoteAddr=r-khorasan.tci.ir:5065
ServerProxy=46.10.123.456:5065
UseProxy=1

What have you tried putting in sip.conf? Note that people here generally don’t know how to use Asterisk based GUIs, so, if Issabel uses a GUI, which I suspect it does, you are in the wrong place.

Outbound proxy should be straightforward. I don’t know what AuthID means, as CallerID and AuthUser use up all the places where SIP could carry that sort of information, and neither of them are specified as containing an “@”, which is not a legal character in a SIP user name.

There is nothing in the information you have provided that indicates you need a type=user section at all. Typically providers only need type=peer, but some providers may need multiple type=peer sections. The only time you would need type=user would be if the provider places their name in the from user field, but it is almost universal that they actually put the caller ID there.

I’ve assumed chan_sip, even though its use is deprecated, because of your references to user and peer, and because I believe that Issabel, may be locked into obsolete versions of Asterisk.