We are having one or two issues setting up a gradwell SIP Trunk with our asterisk PBX and wondered if anyone might be able to shed any light on it.
Basically we have our asterisk PBX registered on gradwells server with no problems, but when we try to dial out we get an error response from gradwell “Forbidden” which related to us not sending our caller ID
allowguest can only appear in the general section.
yes is deprecated as a parameter for nat; use the specific options that you need
username is a deprecated/obsolete name for defaultuser.
Although the default nat= settings are often sufficient for Asterisk inside case, if you have NAT you need to tell Asterisk how to find its external address, so you need to use one of the external address options and/or stunaddr. nat= doesn’t do that; it just applies various hacks to work round the damage caused by NAT, particularly Asterisk outside cases.
More generally, you need to provide sip debug logs, to show where the protocol is getting stuck.
Gradwell replies with 403 Forbidden for some reason - may be caller ID you send is not in the correct format? It would be more helpful if you provided full SIP callflow using debug (sip set debug on from CLI).
NAT misconfiguration could also be the cause of this issue - if you send SIP INVITE with your private IPs in the body, Gradwell could reject them in this manner.