Re: [sipX-dev] [sipx-users] dial plan question
On Fri, 2008-08-15 at 18:25 +0400, Nikolay Kondratyev wrote:
> Damian,
>
> While preparing a text regarding dial rules processing order I made some
> experiments. And it looks like the following algorithm is applied when a
> call arrives:
> Step 1. Internal rules in top to bottom order are processed first. The
> system continues to step 2 anyway.
> Step 2. Then the system tries to match the dialed (possibly modified by
> internal rule) number to configured local extensions and if matched asks
> registrar. The system continues to step 3 only if there was no match on step
> 2.
> Step 3. And finally external rules are processed in top to bottom order.
>
> Is that true?
Close - see
http://sipxecs.sipfoundry.org/rep/sipXecs/main/sipXregistry/doc/Redirection.txt
for a more complete treatment.
> Do I understand right, that only rules, that use gateways, are external?
If a rule goes to a gateway, then sipXconfig puts that rule into
fallbackrules, which means that it is used only if no non-gateway rule
found a Contact.
> I also found, that if dialed number matches a rule, which requires a
> permission, and the user does not have that permission, the rule is matched
> anyway and call fails (my lg-nortel phone shows "temporary failure" in this
> case).
That is correct.
> I think that this behavior is not friendly. I would prefer not to match a
> rule in this case. And let the administrator decide what to do with the
> call. For example, route the call to special AA and play an announcement. Or
> route the call to a different gateway.
Agreed, and I think we have an issue for this, but changing it will be
tricky because we certainly have rules in the field that rely on this
behavior. When (if) we get around to improving this, we'll have to make
allowance for using the "old" semantics.
> This idea, in particular, would allow to implement kind of "user sensitive"
> routing.
> What do you think?
It's been under consideration for a long time (you're not the first
person to bring this up :-) ), but changing it has not gotten into the
must-have list for a release yet.
--
Scott Lawrence tel:+1.781.229.0533;ext=162 or sip:slawrence@xxxxxxxxxxx
sipXecs project coordinator - SIPfoundry http://www.sipfoundry.org/sipXecs
CTO, Voice Solutions - Bluesocket Inc. http://www.bluesocket.com/
http://www.pingtel.com/
