[asterisk-dev] chan_sip SIP Authentication

Jesus Rodriguez jesusr at voztele.com
Thu Jan 29 14:54:09 CST 2009

Hi Olle,

> 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.

Clear now with this and Klaus' explanation. In that case To: header  
was the only way for internal routing on Asterisk but RPID for an  
incoming INVITE should have the caller number, not the called one...  
until i readed the Cisco draft and saw the "called" value for "party"  
parameter :-)



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

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