Hi Brett
On 17/10/12 3:52 PM, Brett Nemeroff wrote:
> Well hold on a sec..
>
> First of all, the TO field is irrelevant. So whatever RURI you have
> (that's the top line INVITE URI), that's where we're sending the call
> to next. If the below invite hits asterisk it should be delivered to
> 111610. If that's not right, you need to set your $rU to whatever you
> want it to be delivered to.
>
This is what the commercial Asterisk provider is saying.
My problem is that the sip config line
"alias_db_lookup("dbaliases","d")" changes the INVITE to our service
number and adjusts the IP address / port all the while losing the
inbound DID so I can not have one registration and a number of phone
numbers.
> Per the docs, the function you are using updates the RURI:
> http://www.opensips.org/html/docs/modules/1.7.x/alias_db.html#id250076
>
> Are you suggesting asterisk is routing on the TO header? This happens
> with some buggy SIP clients from time to time, but I wouldn't expect
> this in Asterisk.
>
No I'm saying that the TO field has the original INVITE with the DID in
it, and because Asterisk DOES NOT routing using the TO field the call is
not correctly routed. (inside the clients Asterisk PBX)
> The "To" Header really shouldn't be considered for routing. That being
> said, there are a handful of UAs out there that insist on doing so. I
> think they are pre-3261 typically but this isn't confirmed.
> -Brett
So I'm asking what should I be doing if I receive a DID INVITE, I need
to route this to a registred using with our losing the ability for the
UAs to correct route the call.
Thanks
Mike
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