Sip gw

I have an Asterisk with is connected to a VOIP provider using an IP authentication. ANd named it as “Ast1”

Now I also created another Asterisk server in a different location and named it as “Ast2”.

I want Ast1 to be the GW for all calls coming from Ast2. I setup an extension on Ast1 and used it as a trunk on Ast2.I am almost making it working but there is still one problem.

When I dial any number, I cannot hear any audio and it disconnects after 30 seconds. However if I dial any number and after 5 seconds I press any number in the keypad, then I can hear the ring or the voice of the opposite party. Forwarded all necessary ports on my router. Is there some other settings needed in box to make this setuo perfect?

Please help

You wouldn’t normally set up an extension, at least in the Asterisk sense, for this, so I’m not clear what you are doing.

It is not clear why the call is failing. The basic failure would suggest inappropriate NAT settings, but the bit about the extra digit makes me want to have the sip debug trace to work out what is really happening.