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.
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.