[asterisk-dev] chan_sip SIP Authentication

[Johansson Olle E]

28 jan 2009 kl. 16.02 skrev Jesus Rodriguez:

> Hi Olle,
>
>
>>>> This implements a way to
>>>> - register with SIp services
>>>> - get the call back
>>>> - match the proper peer, even if you have five accounts, we will
>>>> match
>>>> the proper peer
>>>> - send the call to the called number (to: header), not using a
>>>> pseudo-
>>>> exten that overrides.
>>>
>>> ahh. It took us many yours to tell vendors that To-based routing is
>>> wrong.
>> Oh yes, it is. In theory you should never do that, but...
>>
>> But if you register for a service, the request URI is whatever
>> you register with and can't really be used for any routing decisions
>> in a b2bua.
>> For a simple phone, it doesnt matter. Registration for a trunk  
>> service
>> doesn't really work,
>> which might be a design flaw in SIP.
>>
>> How on earth do you get the requested number without checking To: or
>> RPID?
>
>
> Please, can you explain a little bit more why you need To: or RPID for
> routing decisions? I don't fully understand it :-/
>
> Furthermore, To header is for destination and is always present but
> RPID/PAI makes reference to source information and you don't know if
> they will exist.
>
You only need it in the case where you use the register= configuration  
option
in sip.conf and let Asterisk register as a SIP device to another  
server, mainly with
VOIP service providers. In that case, we register an extension and all  
calls,
regardless of destination will be sent to that extension. If you have  
multiple
DID's, this is of no help, since you want to find out which number the  
call
has as a destination. That informaiton is usually in To: or the RPID.

/O