[asterisk-dev] chan_sip SIP Authentication

Jesus Rodriguez jesusr at voztele.com
Thu Jan 29 14:49:20 CST 2009

Hi Klaus,

>>>>> 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 :-/
> Example: An enterprise has 3 phone numbers.
> +4311111
> +4322222
> +4333333
> The enterprise uses an Asterisk server which registers with the  
> trunking
> provider to receive incoming calls. You do not want the Asterisk  
> server
> to register for each number - Asterisk registers only once and the
> trunking provider maps all these numbers (or more) to the single
> registration.
> Usually SIP routing is based on the RURI. But in this case, the RURI
> received by Asterisk is usually the Contact URI of the REGISTER.
> Thus - how does Asterisk find out the originally called number?
> A Workaround is to put the originally called number in the To header -
> but this is ugly as To based routing it is against all RFCs.

Ok, clear now... been there with that problem and yes, we had to use  
the To header for sending the incoming call to the right Asterisk  



Jesus Rodriguez
VozTelecom Sistemas, S.L.
jesusr at voztele.com
Tel. 902360305

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