[asterisk-users] SIP Registration and INVITE question

Steve Davies davies147 at gmail.com
Tue Apr 7 05:08:28 CDT 2009

2009/4/7 Olle E. Johansson <oej at edvina.net>:
> The REGISTER request in the RFC was really written for a device.
> The way providers use it for trunks with multiple DIDs is outside of the
> RFC and is discussed in relation to the SIPconnect specification in
> the SIP forum.
> Some providers solve this by not using the Contact: in the register
> request at all for the calls, instead guessing a URI with the DID
> in the user name part, something that breaks communication
> even more as the Contact might include other hints on call routing
> internally, like line button in a SNOM phone.
> I would say that the only way right now is to parse the To: header.
> I started working on some code a while ago that would handle
> this better, but never completed it. We simply registered a random
> string and then replaced it with whatever was sent in the To: header
> (which should be the original destination) before hitting the dialplan.
> That code still exists in a branch somewhere and in Pineapple.
> This code would also solve the issue with registering multiple
> accounts with one provider.
> /O

Thanks Olle, as always, a useful response :)

In the meantime, I suspect  that the following is the current dialplan
based workaround for calls that come in to 's' because of a default
Registration Contact?

exten => s,1,Set(DN=${SIP_HEADER(TO):5})
exten => s,n,Set(DN=${CUT(DN,@,1)})
exten => s,n,GotoIf($["${DN}" = "s"]?:default,${DN},1)
exten => s,n,Hangup()

Comments or improvements anyone?

Thanks again.

More information about the asterisk-users mailing list