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I have a call forwarding service setup to forward calls made to a foreign number to our on-premise 3cx (v20) server via SIP address. I am getting this error and not sure where to go with it... It seems it has to do with inbound call settings not being correct, but not sure where in 3CX to make changes or inspect this further.
Unidentified Incoming Call. Review INVITE and adjust source identification: INVITE sip:805@**OURIP** SIP/2.0 Via: SIP/2.0/UDP 66.110.73.75:5060;rport=5060;branch=z9hG4bKPj6ce25e2a-9f43-4e1d-b269-b309d93b85e5 Max-Forwards: 70 Contact: <sip:[email protected]:5060> To: <sip:805@**OURIP**> From: "+2347001235382" <sip:[email protected]>;tag=72c16ad1-557d-4e85-b31e-1a1a5fb503bf Call-ID: e873291d-659c-4243-b4b3-47c129ae5044 CSeq: 18130 INVITE Session-Expires: 1800 Min-SE: 90 Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Content-Type: application/sdp Supported: 100rel, timer, replaces, norefersub, histinfo User-Agent: Asterisk PBX 18.10.0 Diversion: <sip:[email protected]>;reason=unconditional Content-Length: 282 v=0 o=- 944535616 944535616 IN IP4 66.110.73.75 s=Asterisk c=IN IP4 66.110.73.75 t=0 0 m=audio 33108 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv
Unidentified Incoming Call. Review INVITE and adjust source identification: INVITE sip:805@**OURIP** SIP/2.0 Via: SIP/2.0/UDP 66.110.73.75:5060;rport=5060;branch=z9hG4bKPj6ce25e2a-9f43-4e1d-b269-b309d93b85e5 Max-Forwards: 70 Contact: <sip:[email protected]:5060> To: <sip:805@**OURIP**> From: "+2347001235382" <sip:[email protected]>;tag=72c16ad1-557d-4e85-b31e-1a1a5fb503bf Call-ID: e873291d-659c-4243-b4b3-47c129ae5044 CSeq: 18130 INVITE Session-Expires: 1800 Min-SE: 90 Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Content-Type: application/sdp Supported: 100rel, timer, replaces, norefersub, histinfo User-Agent: Asterisk PBX 18.10.0 Diversion: <sip:[email protected]>;reason=unconditional Content-Length: 282 v=0 o=- 944535616 944535616 IN IP4 66.110.73.75 s=Asterisk c=IN IP4 66.110.73.75 t=0 0 m=audio 33108 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv
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