Creating another SIP trunk with different DID number but under same provider

[imalaysian]

Hi 3cx forum, beginner here in setting up SIP trunk.

I am having some trouble configuring multiple SIP trunks that are under the same provider. I wanted to call the “DID” number and have it route the call to IVR/Digital Receptionist.

The weird problem now is that I am able to call the “DID” number of the first trunk and route successfully to the IVR. However, I am unable to call the “DID” number of the second trunk and route to IVR. It just shows up forbidden like this
Call to T:Line:10000>>0397797876@[Dev:sip:[email protected].174] from L:14.1[Extn:102] failed, cause: Cause: 403 Forbidden/INVITE from xxx.xx.xx.174:5060


I have also analyzed the history log. It appears that the “FROM”, “TO”, and “CONTACT” are very different.


I have tried the Inbound Parameters solution seen in other threads by checking/unchecking “Call source identification” and the bottom check boxes as well. I have also checked the outbound rule and it does not seem to be that issue.
https://markuphero.com/share/tTTw5TnY0Occi95md1z5

I am using Generic SIP Trunk and it is register/account based type with IP.

Any advice?

 

[YiannisH_3CX]

Hi @imalaysian

Please note that I deleted your screenshots as they contained some sensitive information you do not want public. From what I saw however you were comparing Invites from different sources. The first was from the PBX to a client via the tunnel and the second was from your provider. Of course these would be different so make sure you compare the same things in the future.
My first suggestion would be to use a supported 3CX provider to avoid all the guessing work.
The next step is to use only one trunk from the same provider but with multiple DIDs if possible.
If the 2 suggestions above are not possible then here is the documentation on how the PBX matches calls to SIP trunks so you can adjust your settings accordingly. https://www.3cx.com/docs/sip-trunk-inbound-calls/

 

[imalaysian]

Thanks for the reply. I managed to solve it by reconfiguring the outbound parameters and also just using 1 SIP trunk for now. However, now I have another problem regarding on one way audio issue. I have run the the firewall checker test, status all return fine except the SIP alg. I not sure how to disable this setting as it is not available in my router. My router is a china brand router Xiaomi AX6000.

Things I have tried
-Fresh backup the 3cx
-Changed the Network from static to dynamic (vice versa)
-Configured various setting: PBX deliver audio, support reinvites
-Used wireshark to capture packets but I don't have a clear idea on what I am looking at but i mostly look at these 2 RTP/SIP protocol

 

[cobaltit]

So you have an obvious error with the SIP ALG so rather than doing all those other things I would focus on resolving that. This would require reaching out to the router manufacturer or the internet provider if it's their router, or replacing the router if it's yours. Or using a cloud installation where SIP ALG won't matter.