SIP trunk problems

Hi guys. I’m newby and i’m going crazy trying to install my SIP trunk.
I have one asterisk PBX installed and 4 internal extensions (3 inside my NAT and 1 outside the NAT). All is working fine.
Before asterisk i have one SIP provider for pstn CALLS for my sipura SIP phone or my X-lite softphone. All working fine also.
Now i want to have the asterisk PBX as a client for my SIP provider and make calls from my extensions.
I made the SIP trunk but always i have the same message. I debugged the problem with the asterisk console and i get:
"==everyone is busy/congested at this time (1:0/1/0)
– Executing [1;36;40…
– Goto (macro-dialout-trunk,s-CONGESTION,1)"
My hardware is simple: 1pc asterisk installed, 1 router netgear, 1 sipura and 2 pc’s running x-lite.
I don’t understand why the x-lite or the sipura are working fine directly over my SIP provider but when i try to connect from asterisk i have this message. Of course i’m registered. If i try “sip show registry” i get "…sip provider registered"
Any idea?
thank you in advance.

Why don’t you simplify for your extensions.conf to see what happens.

Maybe your statements are wrong?

Maybe use ethereal to check the sip messages?


PAtrick Arkley

Hi parkley, thank you for your answer. But i’m really going crazy. I have 2 providers, one is going fine and the other one is a problem. I read te sip packtes from the X-lite and from Asterisk and i read the same , just i ’ don’t know what more to do.

you could post the relevant parts from your config and a debug log file fragment for a failed call ??

Hi buconbuttie, thanks for your answer
here is the link to the file that you ask me for. … solrac.txt

The file is a bit long, sorry about it. I put in the file:

X-ten lite log (this one is long but was the debug file from xten for the registration and the call)
the log from Asterisk console

If you can help me i’ll apreciate it.
thanks in advance.

Someone read the logs? I someone have an idea please tell me.
thank you :cry: :cry: