[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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