[Asterisk-Users] SIP and NAT - best practices?

[Trevor G. Hammonds]

Leo Ann Boon <> wrote on Sunday, 22 January 2006 4:32 PM:

> Trevor G. Hammonds wrote:
> 
>> While I have not used siproxd, I have read a bit about it.  From my
>> understanding of the docs, the local SIP agents register to siproxd,
>> but siproxd does not register to Asterisk.  So the calls will
>> traverse 
>> the NAT properly, but features like MWI will not work in this
>> scenario. Also, this would be pure SIP URL dialling (e.g.
>> usernam at domain.com) as opposed to traditional telephone dialling
>> (e.g. 1-213-555-8080). 
>> 
>> Please correct me if I am wrong, because I would really like to be
>> (in this case).  :-) 
>> 
>> 
> The docs are a little confusing. Look in the FAQ section: What types
> of operation does siproxd support? 
> Here's the text.
> 
>>   1) Siproxd as outbound proxy:
>>      - Configure your local client to register with some 3rd party
>>        service like Sipphone, FWD, Sipgate or any other.
>>      - Configure your local client to use siproxd as OUTBOUND PROXY
>> 
>>      Note: In this case, the local client does NOT register with
>>      siproxd but only with the external SIP restration service. The
>>      only condition is that siproxd needs to stay in the path of
>>      communication, therefore the local client must be configured as
>> to use an OUTBOUND PROXY. 
>> 
> That's all you need to do. All your clients will still register to
> Asterisk through siproxd, siproxd will take care of rewritting the
> SIP headers to differentiate requests for each client.  
> 
> Leo

Thank you, Leo!  This is exactly what I need.  I am going to play around
with that really soon.

		Trevor