On 3/4/10 11:04 AM, Tony Graziano wrote:
This is one aspect of:


    http://track.sipfoundry.org/browse/XX-6358

    and should clearly be fixed.

In the meantime I use this rule:

Create a custom rule, give it local permissions. Prefix is "411" and "0" digits, resulting call is

800-466-4411 and append nothing, choose the gateway and activate it after moving it to the appropriate place in your dialplan.

It does work.

but I don't want to dial a number through an ITSP. I want it to make a sip call. dialing through a ITSP uses outbound mins and bandwidth. making a sip call only uses bandwidth.

but this is a bug or not? is this intended behavior? because if you validate 'SIP URL' then it must accept: (or am I wrong?)

   * abc (might be someone's alias)
   * 123 (might be an extension, or an alias)
   * [email protected]
   * [email protected]
   * aaannnp...@ipaddress

maybe I need a virtual user, and call fwd always to the sip url.
if @domain.com (or ip address) matches THIS sip domain or ip, it would be an internal call. if not, and you have internet dialing NOT going to sipxbridge, it should just make the same sip call as if you dialed the url on the phone (I would much rather dial 441 on the phone than add this to everyone's speed dial)

(i just set up a virtual user, 411, with alias 8005551212 and 18005551212, 8004664411 and 18004664411 call fwd all to [email protected]
Call FWD validation box accepts a SIP URL.

so, 4.2x.. will we be able to do this with a dial plan?

I think if the text box says it takes a SIP URL, it should take a SIP URL, in all that that means.
If the text box says extension, number or alias, it could exclude @.

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Michael Scheidell, CTO
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