[OpenSIPS-Users] "Asterisk Contexts" in OpenSIPS

[Deon Vermeulen]

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.


Regards
Deon


On 01 Oct 2010, at 8:42 AM, Raúl Alexis Betancor Santana wrote:

> On Viernes 01 Octubre 2010 06:11:48 Deon Vermeulen escribió:
>> Pardon for asking but how do I do this?
>>
>> Don't understand what you mean by "dump your SIP traffic and make  
>> sure
>> that you route the ACK message to the correct destination" or how to
>> do this.
>
> It means using tshark, ngrep, tcpdump or witchever traffic capture  
> tool you
> feal confortable with
>
> Best regards
> -- 
> Raúl Alexis Betancor Santana
> Dimensión Virtual
>
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