Unable to get incoming calls, external calls work fine.

[GerardW_0123]

When I try to call my PBX through my number I get the response that the number isn't currently available. It works normally when I call from my PBX to other numbers. My PBX is hosted by 3CX.

This is the error I'm getting:
06/20/2022 4:46:44 PM - [CM102001]: Authentication failed for AuthFail Recv Req INVITE from 188.94.184.98:5060 tid=082f.f4755d734d63ecddade9b2e73d324e22.0 Call-ID=14f8c738-5d47-4113-aa72-eaff1c604bb8: INVITE sip:+PBX [email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 188.94.184.98;branch=z9hG4bK082f.f4755d734d63ecddade9b2e73d324e22.0 Via: SIP/2.0/UDP 188.94.184.104;received=188.94.184.104;rport=5060;branch=z9hG4bK0XervQrSZDSDp Max-Forwards: 66 Record-Route: <sip:188.94.184.98;lr=on;ftag=8NS48y680y8NQ;did=40a.b941> Contact: <sip:[email protected]:5060;transport=udp;gw=vp-i10router01> To: <sip:+PBX [email protected]:5060> From: "EXTERNAL NUMBER"<sip:[email protected]>;tag=8NS48y680y8NQ Call-ID: 14f8c738-5d47-4113-aa72-eaff1c604bb8 CSeq: 53314411 INVITE Session-Expires: 1800;refresher=uac Min-SE: 90 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY Content-Disposition: session Content-Type: application/sdp Proxy-Authorization: Digest username="FreeSWITCH",realm="3CXPhoneSystem",nonce="414d535962b0885497:613889320125bd0137b1072b835fa2bc",algorithm=MD5,uri="sipBX [email protected]:5060",response="f9627d747789f6351d8e338fbe4b2750" Supported: timer, path, replaces User-Agent: CMSBC Allow-Events: talk, hold, conference, refer Privacy: none P-Asserted-Identity: <sip:+EXTERNAL [email protected]> Content-Length: 267 v=0 o=CMSBC 1655710154 1655710155 IN IP4 188.94.184.109 s=CMSBC c=IN IP4 188.94.184.109 t=0 0 m=audio 50756 RTP/AVP 8 101 a=silenceSuppff - - - - a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:50757 a=ptime:20 ; Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings

I hope someone can help me.

 

[Alphabetic]

How many SIP trunks have you got?

Does your SIP trunk that you're calling in to show as registered?

 

[Saqqara]

What SIP provider, maybe worth calling them to see if the routing is correct at their end.

 

[GerardW_0123]

I have two SIP trunks, both show as registered.
I have tried calling my SIP provider, but they say the routing is correct at their end.

 

[Alphabetic]

In your outbound rule, i assume you've set up the 2 SIPs so when you dial out, if it fails on one its going out on the other.

you should to do a wireshark capture to see a) if the call comes in b) invite format

are you using a supported SIP provider?

 

[GerardW_0123]

Yes, I did set it up like that, but my outbound is fine.
I am using a supported SIP provider, cm.com.

How do I do a wireshark capture?

 

[Alphabetic]

You need to disable one of your SIPs otherwise you're going to get confused. You need to test each one of your SIPs at a time to find which is the one at fault otherwise youre chasing your tail.

What SIP does your number come in out? that is the one you want to keep enabled.

If your call goes out fine on this sip then you need to look at inbound rules to make sure that you have one that matches your number.

https://www.3cx.com/docs/capture-network-traffic/

 

[GerardW_0123]

Currently, the call goes through, but it's not getting sent to my phone.
It worked before, but maybe I missed a setting.