[asterisk-dev] chan_sip SIP Authentication
klaus.mailinglists at pernau.at
Thu Jan 29 01:52:57 CST 2009
>>> How on earth do you get the requested number without checking To: or
>> Using To: you assume that the To header contains the originally called
>> number. But that depends on the setup of the trunking provider.
>> (what if
>> the trunking provider uses Asterisk?)
> Yes, one has to check with SIP debug first. Always. And if the trunking
> provider use Asterisk I've given him a dialstring option to set the to:
> header properly - which you of course have seen, since you've read
> the docs, right? ;-)
Which document describes the dialstring of the SIP channel?
>> Yes you are right - SIP trunking is not specified somewhere.
>> The trunking provider I use maps the called number in the RURI (thus
>> ignores the userpart in the REGISTER contact but just uses the
>> domainpart for routing). Of course this would cause problems if you
>> register twice to a trunking provider and has do differ incoming call
>> (which is done on the RURI).
> Talk about not being SIP-compliant :-) But yes, I've seen that too.
> Which shows that something is missing here.
>> Isn't there a dedicated P-.... header specified in IMS to signal the
>> originally called number?
> Maybe. I've used RPID with party=called in some cases.
Found it: http://www.ietf.org/rfc/rfc3455.txt
P-Called-Party-ID (there is is good documentation why other workarounds
are not good).
This remembers me on another approach - differentiate between routing
and retargeting. Please read
also addresses these problems. As this is logically not retargeting but
routing, the trunking provider could send the called number in the RURI,
and the registered contact is sent as Route header. Thus, Asterisk could
match the registered contact against the topmost Route header and the
RURI contains the dialed number.
Probably, as there are multiple solutions, we should ask on the SIP
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