[asterisk-users] Error redirecting an incoming call of a SIP provider to a local extension

[Benoit]

On 20/02/2010 01:35, Daniel Bareiro wrote:
> alderamin*CLI>
>      -- Executing [300 at from-internal:1] Dial("SIP/danib-089f8820",
> "SIP/300|30|tTrm") in new stack
> [Feb 19 19:22:50] WARNING[19254]: app_dial.c:1237 dial_exec_full: Unable
> to create channel of type 'SIP' (cause 20 - Unknown)
>    == Everyone is busy/congested at this time (1:0/0/1)
>    
Well, looks like your * server is simply unable to dial the sip user '300'.
  Either there is some call-limit in place, or problem with the registration
of the phone ?

> It is probable that this can be due to a problem of interaction between
> contexts? I copy the content of extensions.conf and sip.conf to see if
> it can help to find the problem:
>    
What could be of some use, is the result of "sip show peer 300"