SIP Trunk - 404 Error

[ccc]

Hello,

I'm trying to implement 3CX PBX in my company. Everything working good, the only problem is a specific SIP Trunk that is giving a huge headache.

With Asterisk 13 using SIP, I'm able to connect(receive/make calls) to my trunk with the following configuration:

[my-trunk]
type=peer
context=my-trunk-context
host=my-trunk-ip
qualify=no
disallow=all
allow=g729,ulaw
nat=force_rport,comedia
dtmfmode=rfc2833
insecure=port,invite
call-limit=100

But with 3CX, I create a SIP Trunk, specifying:
-Host
-Type of Authentication: Do not require
-Main Trunk No: trunk-number

I also create the route out and in, but I call to somewhere it gives me "host-ip replied: 404 No Routes Found; from IP: host-ip". I can receive calls but it muted.

I don't know what to do more, this post it's my last chance.

Thank you so much.

 

[cobaltit]

If the 404 is coming from your trunk provider, that's who is rejecting your call and that's who you need to talk to. They can tell you why the call is being rejected

 

[eddv123]

To add a little further on what @cobaltit has said already, a 4xx style response often denotes a client-side issue (your trunk configuration basically) which is causing the provider to reject the call.

I would assume in this case you are using a 3CX "Un-supported" provider (which does not include an in-built system template) and thus requires tweaking for their settings - as already stated however it is down to the provider to tell give you this information.

 

[JohnS_3CX]

If you can receive calls but no audio, then you need to check your codec first, and also run the firewall checker. The PBX will need to have some open ports in order to pass audio.
https://www.3cx.com/docs/ports/

As for outgoing calls you will need to ensure you have an outgoing rule that actually matches the destination you called.
https://www.3cx.com/docs/manual/sip-trunks/#h.3evqdta9sabf

 

[simply7]

Any chance this trunk has a separate route that requires a specific NAT rule? It sounds like your call setup is good but your RTP is not working. A good wireshark trace should help.