Self hosted 3CX - Outbound calls work fine, Inbound calls do not work at all with Gradwell?

[Korallis]

As the title says,
i have outbound working fine however inbound calls do not work at all ..
and i cannot figure out what ive done wrong.
the servers running on a windows Server,
the provider sent me the call routing information and told me "your fault is with your PBX"

 

[saint_]

Make sure you have all of your ports (firewall) setup correctly.
Also, some service providers have something called SIP ALG in their router, which at the end causes issues if you use your own PBX. Make sure your internet router does not have it, and if it does, turn it off.

 

[AWS2P]

How did you connect your phones ? Not in STUN I hope as in 3CX hosted only supported way is SBC

 

[Korallis]

How did you connect your phones ? Not in STUN I hope as in 3CX hosted only supported way is SBC

i have some in stun. but i also have the App on my laptop which auto provisioned without stun.

the issue isnt the phones, the issue is with the pbx itself.

 

[Korallis]

Make sure you have all of your ports (firewall) setup correctly.
Also, some service providers have something called SIP ALG in their router, which at the end causes issues if you use your own PBX. Make sure your internet router does not have it, and if it does, turn it off.

PBX passed all firewall rules..

 

[AWS2P]

firewall test is pbx side , the one important is on remote location (where you use the phones)

App is not connecting same way , just using tunnel on port 5090.

What happens exactly on inbound call? do they almost reach the PBX?
Did you set activity log in verbose to increase detail level.
Do you have set one inbound rule ?
If so can you make a screen share of that rule and hide sensitive infos.

Remove your previous screen copy as you let all sensitive info visible. (this is a public forum)

 

[Korallis]

firewall test is pbx side , the one important is on remote location (where you use the phones)

App is not connecting same way , just using tunnel on port 5090.

What happens exactly on inbound call? do they almost reach the PBX?
Did you set activity log in verbose to increase detail level.
Do you have set one inbound rule ?
If so can you make a screen share of that rule and hide sensitive infos.

Remove your previous screen copy as you let all sensitive info visible. (this is a public forum)

Sorry im kinda new to this.
the provider said they reach the pbx but then nothing happens.
the inbound rule is attached.

this is what happens when i call

09/09/2020 10:35:54 - [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification: Invite-UNK Recv Req INVITE from 109.224.###.###:5060 tid=af1d.0796a7a2.0 Call-ID=4515e0000ef5-5f58a1fa-393cd902-1b88bc90-1756ae70@127.0.0.1: INVITE sip:441212856###@91.142.##.### SIP/2.0 Via: SIP/2.0/UDP ###.###.###.###;branch=z9hG4bKaf1d.0796a7a2.0 Via: SIP/2.0/UDP ###.###.###.###:5060;branch=z9hG4bKaf1d.e554ad8.0 Via: SIP/2.0/UDP ###.###.###.###:5060;branch=z9hG4bK00E0F5150E45D30665FF8A139443 Max-Forwards: 68 Record-Route: <sip:###.###.###.###;lr=on;did=2a.2cfe6174> Record-Route: <sip:###.###.###.###;lr;ftag=0UUHS0000030000E1D01001u0X675A71UF1C70;did=2a.3666e5e2> Contact: <sip:###.###.###.###:5060> To: <sip:44121######@###.###.###.###> From: <sip:0789######@###.###.###.###>;tag=0UUHS0000030000E1D01001u0X675A71UF1C70 Call-ID: [email protected] CSeq: 1136 INVITE Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE Content-Type: application/sdp Supported: timer, replaces, histinfo, 100rel User-Agent: TELES-SBC Allow-Events: refer P-Asserted-Identity: <sip:[email protected]> Session-ID: 1599644154.c8cd6a660ad6e12a Content-Length: 463 v=0 o=- 2112501657 0 IN IP4 194.145.191.142 s=TELES-SBC c=IN IP4 ###.###.###.### t=0 0 m=audio 8070 RTP/AVP 8 0 18 97 2 99 4 80 96 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 G726-16/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:99 G726-40/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no;bitrate=6.3 a=rtpmap:80 G723/8000 a=fmtp:80 annexa=yes;bitrate=5.3 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15

 

[AWS2P]

Try to change your inbound rule like this:
Change format number in DID/DDI field to *2856393
After validation , do a call test ,extension 001 should ring.

 

[Korallis]

Try to change your inbound rule like this:
Change format number in DID/DDI field to *2856393
After validation , do a call test ,extension 001 should ring.

You sir are a legend.
that worked perfectly.
The gradwell Guide clearly tells you to put +44 and not * lol changing to * worked perfectly thank you kindly

 

[AWS2P]

Problem was format number sent by provider was not exactly same in your rule so it doesn't match.
now starting with * your rule just wait to match with what's after the *