Nothing but 50029 Events on all my calls

TPR3CX

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Greetings,

I just upgraded to V20 from V18. It's still self-hosted and moved it to an AWS EC2 server. For the EC2 Security Group we allow all outbound traffic and for inbound we allow:
TCP
80, 443, 5000, 5001, 5015, 5060, 5061, 5090
UDP
5060, 5090, 9000-10999

My SIP provider is Flowroute.

All calls coming in seem to be coming in with event 50029 with details being, Call from "LAST NAME FIRST NAME" <sip:[email protected];isup-oli=00>;tag=gK00669b7f to <sip:[email protected]> has been dropped as it was not acknowledged in time. Please verify configuration and ensure there are no connectivity issues.

I'm totally baffled on what the issue is.
 
Hi,

Is your firewall check Green? Can you switch on Flowroute (and 3CX) to use TCP for SIP/RTP and not UDP as AWS can have routing issues with UDP.

Best,
KS.
 
And please close all the unneccessary ports... You only need the https port of your pbx, if its 443 than you can close 80, 5000,5001,5015
 
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Hi,

Is your firewall check Green? Can you switch on Flowroute (and 3CX) to use TCP for SIP/RTP and not UDP as AWS can have routing issues with UDP.

Best,
KS.
My Firewall is green. Passes all the tests. Not sure where those settings are on Flowroute and 3CX.
 
And please close all the unneccessary ports... You only need the https port of your pbx, if its 443 than you can close 80, 5000,5001,5015
Will do.
 
OK I think we're getting close. I changed the transport to TCP on the FlowRoute side. Not sure how to force that on the 3CX side. Now it's a 30051 error.

Unidentified Incoming Call. Review INVITE and adjust source identification:. INVITE sip:[email protected];transport=tcp SIP/2.0. Via: SIP/2.0/TCP 34.226.36.32:5060;branch=z9hG4bK4f7a.c0ce095d4d8f5d87f7907d437695d5cd.0. Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bK4f7a.50b5619b1c1d83b55d21bfc4059fcc60.2. Via: SIP/2.0/UDP 34.211.68.10:5060;branch=z9hG4bK4f7a.be3d9b7a8461f866ed41369b247f5eca.0. Via: SIP/2.0/UDP 68.64.80.114:5060;branch=z9hG4bK00Befe293c667c18fed. Max-Forwards: 66. Record-Route: <sip:34.226.36.32:5060;transport=tcp;r2=on;lr>. Record-Route: <sip:34.226.36.32;r2=on;lr>. Record-Route: <sip:34.211.68.10;lr>. Contact: "NEW YORK"<sip:[email protected]:5060>. To: <sip:[email protected]>. From: "aaaaaaaaaaaaa" <sip:[email protected];isup-oli=00>;tag=gK00021cf6. Call-ID: [email protected]. CSeq: 682717 INVITE. Session-Expires: 1800. Min-SE: 90. Content-Type: application/sdp. Supported: timer. P-Asserted-Identity: "xxxxxxxxxxxxx" <sip:+1xxxxxxxxxx;verstat=[email protected]>. P-Charge-Info: tel:xxxxxxxxxx. Content-Length: 218. v=0. o=- 383198 29448 IN IP4 68.64.80.4. s=-. c=IN IP4 68.64.80.4. t=0 0. m=audio 1482 RTP/AVP 0 8 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=maxptime:20

According to the 3CX event log docs, this is a 30051 error.

30051Unidentified Incoming Call. Review INVITE and adjust source identification:
%1.
WarningThis is a problem when the Source is not trusted.
3CX expects SIP call traffic to come from a certain IP Address and with a certain format and fails to satisfy 3CX checks. For security purposes 3CX will not allow this request and show this message.

So What do I need to change on my 3CX machine to get this work finally work.
 
Is the DID set in that format in the trunk?
 
Is the DID set in that format in the trunk?
Not sure what you mean by that. For the main number I had it as 1xxxxxxxxxx and +1xxxxxxxxxx. Both yielded the same results
 

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