Error with outbound call

[erdia]

Good afternoon, I have correctly configured a PBX with SIP extensions and trunks, and receive calls and everything, but when you call me out I get an error "404 Not Found / INVITE from xxx.xxx.xxx:5060" and will not let me call . Exit rules I set the Prepend I of 0 and is removed when dialing. Help! Thank you

 

[Sopock]

I've also changed the ports of the PBX because router when VoIP lets not map the port 5060. Then change it to port 6500 and redirect it to my PC. (192.168.1.103).
http://i68.tinypic.com/288qxpc.png

why is that even needed in this case when providers proxy is available from local network?

5060 Yes – if you intend to use VoIP Providers, WebRTC and Remote Extensions that are NOT using the 3CX Tunnel Protocol

9000-9255 Yes – if you intend to use remote extensions or VoIP Provider

 

[erdia]

I've also changed the ports of the PBX because router when VoIP lets not map the port 5060. Then change it to port 6500 and redirect it to my PC. (192.168.1.103).
http://i68.tinypic.com/288qxpc.png

why is that even needed in this case when providers proxy is available from local network?

5060 Yes – if you intend to use VoIP Providers, WebRTC and Remote Extensions that are NOT using the 3CX Tunnel Protocol

9000-9255 Yes – if you intend to use remote extensions or VoIP Provider

No, I don't have any proxy in my local network. 404 error keeps coming and still do not know why :roll: :roll: Thank you all people follow my post!

 

[Sopock]

No, SIP proxy server on Livebox is part of your LAN. You can register several softphones on that server without a need for port mapping. 3CX may continue to use 5060 port.

But first, to determine right length <sip:[email protected]:5060> use any softphone with comprehensive logging such is X-Lite registered to 192.168.1.1 :
Optionally, click Help > Troubleshooting and test your devices and audio quality
click on Support tab to view log :?:

 

[erdia]

No, SIP proxy server on Livebox is part of your LAN. You can register several softphones on that server without a need for port mapping. 3CX may continue to use 5060 port.

But first, to determine right length <sip:[email protected]:5060> use any softphone with comprehensive logging such is X-Lite registered to 192.168.1.1 :
Optionally, click Help > Troubleshooting and test your devices and audio quality
click on Support tab to view log :?:

Recording directly to Livebox softphone, if you let me dial and call and everything is OK, but not from 3CX. Equally touching the Livebox have found the SIP data that is connected and configured with 3CX and I now have no line Livebox and calls go directly to 3CX, but call me appears "403 Forbidden invite from sip.orange.es : 5060 "Another problem D ': Error 404 solved finding the SIP data directly, but now 403. Thanks!

25-jul-2016 09:37:01.496 [CM503003]: Call(C:1): Call to <sip:[email protected]:5060> has failed; Cause: 403 Forbidden/INVITE from 85.62.240.79:5060
25-jul-2016 09:37:01.353 Call to T:Line:10000>>93XXXXXXX@[Dev:sip:[email protected]:5060] from L:1.1[Extn] failed, cause: Cause: 403 Forbidden/INVITE from 85.62.240.79:5060
25-jul-2016 09:37:01.262 [CM503003]: Call(C:1): Call to <sip:[email protected]:5060> has failed; Cause: 403 Forbidden/INVITE from 85.62.240.79:5060
25-jul-2016 09:37:01.198 Call to T:Line:10000>>1004@[Dev:sip:[email protected]:5060] from L:1.1[Extn] failed, cause: Cause: 403 Forbidden/INVITE from 85.62.240.79:5060

 

[Sopock]

Why is manual configuration sometimes necessary?

in outbound parameters you can see that it requires advanced SIP knowledge :!:

• packet capture
• Advanced Custom Gateway and VoIP Provider Configuration