Hello everyone,
I want to set a sip trunk tunnel between 2 Asterisk servers. Can anyone recommend a good resource to learn from?
Asterisk (and the SIP RFC) do not have a concept of trunks. For what you seem to want to do, treat each side as though it were a provider, and don’t override the From header. Providers generally don’t authenticate themselves to users, but there is no reason why you shouldn’t authenticate both ways with either the same, or different, secrets.
I see, thank you david, in that case, do you know some good resource to learn how to do this?
It is very easy once you understand Asterisk. Too much Asterisk configuration is done without a proper understanding, and I don’t want to further encourage that. Basic provider configurations are given in res_pjsip Configuration Examples - Asterisk Project - Asterisk Project Wiki but this is over-engineered for your purposes, as you don’t need registration and you can start without authentication.
(I suspect copied and pasted configurations are going to be replaced with ChatGPT ones 
.)
Dont worry David,
i did not use ChatGPT, i am gonna look into what you ve send me
Company that I am working for is this small town tech company and so far, they hadnt decided whether they are gonna convert from sip to pjsip so my examples are in sip.
extensions.conf
sip.conf
so far I have smth like this, this is on one server, second one looks almost identical but host is different.
When I try to call I get this:
Any idea what did I miss?
Please use plain text, not images.
You don’t have a peer definition for 6003.
Also calls to 6003 and 6004, will fallback to calling SIP/6003@trunk and SIP/6004@trunk if SIP/virtual-[AB] fails, although the Noop will not be executed in those cases. This is not what your log shows, so I don’t believe your dialplan is the actual one in use.
It’s easy. You can create SIP trunk Tunnel between the servers using simple configuration .
Yes, later I figured, it out. Right now its working, thank you for your patience with me, you are amazing, David.
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