[OpenSIPS-Users] Inbound DID TO/INVITE issue
flavio at asteriskguide.com
Wed Oct 17 13:15:33 CEST 2012
One of the best ways to solve it is to include the DID in an extra header.
We usually include something such as X-DID: did_number. Then in the
Asterisk/Freeswitch PBX server you can recover this DID reading this header
and routing internally. Many ITSPs implement in this way. Thus, you can
have a single registration but multiple DIDs.
Flavio E. Goncalves
2012/10/17 Kevin Sandy <kevin at digitallotus.com>
> Then the other providers are doing something similar to the prior
> suggestion - replacing $rU with $tU before relaying.
> One registration = one contact URI. So, multiple registrations, rewrite
> the user, or ditch the registrations and route to them by IP or host name.
> kevin sandy, dcap, mcp
> On Oct 17, 2012, at 6:49 AM, Mike O'Connor <mike at oeg.com.au> wrote:
> > Hi Kevin
> > Surely there is a better solution than this, because all the Asterisk
> > systems I've seen have inbound routing without a registration for each
> > The supplier of the commercially support Asterisk would need to make
> > changes which they are this point are not prepared to support, when
> > every other ITSP they have connected to in the USA and Europe, has
> > supported the one registration and more than one DID.
> > So I really need to understand what the standard should be and how my
> > config is incorrect.
> > Cheers
> > Mike
> > On 17/10/12 8:54 PM, Kevin Sandy wrote:
> >> We resolved a similar issue by having Asterisk register once for each
> DID. Alternatively, you could modify their dial plan to take the To header
> into account, but I'd go the multiple registrations route if possible.
> > _______________________________________________
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