[OpenSIPS-Users] "Asterisk Contexts" in OpenSIPS

Raúl Alexis Betancor Santana rabs at dimension-virtual.com
Fri Oct 1 09:22:43 CEST 2010


On Viernes 01 Octubre 2010 07:53:38 Deon Vermeulen escribió:
> Hi Raul
> 
> Thanks for the clarification and response. Really appreciate it.
> 
> Have been looking at the siptraces provided by SIP Trace in Opensips
> Control Panel.
> 
> I'm guessing I still have a NAT Traversal issue.
> 
> What is really strange is that I can only phone from usera at domaina.com
> to userb at domain.com, but not visa-versa.
> When I answer the call on userb at domain.com the call does not setup but
> times out with error 408 on both ends.

If as I suppose, you are new to OpenSIPS, I suggest you to begin with the 
standar config file, it does nat-fixing-handling, and when you undestand what 
it does, try to modify it for adding what youe need.

Also a bunch of SIP knowleadge is "a must".

Best regards
-- 
Raúl Alexis Betancor Santana
Dimensión Virtual



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