[Asterisk-Users] Dial via sip gateway?

[Rich Adamson]

> > From what I can tell (box is about 48 hrs old for me), it 
> > seems to be a rather incomplete or just-bare-sip-minimum 
> > functionality. It also appears as though all four ports are 
> > treated as a group-of-lines, and one doesn't have any choice 
> > (from a sip perspective) on which port to use for outgoing 
> > calls. Since this one is set up with 1:home, 2:business, 
> > 3:outgoing calls I really need to be able to select which 
> > port * is going to use, particularly since outgoing 'home' 
> > long distance calls must use a different port then for 
> > outgoing 'business' calls.
> 
> I have an idea of a crude hack that just might work - e.g. if you need
> to dial a number on line 3, first make two outgoing calls to a bogus
> number (just to keep the lines busy for a second) and then place the
> 3rd call to the destination you want - if I understand the situation
> correctly, the 1204 should dial on the 3rd line then and the first two
> calls should drop quickly (no such number). Of course, in that case
> you need to keep the line state e.g. in the DB so that, say, line 1 in
> use doesn't mess things up.
> 
> Yes, I know it's ugly. If it's also bound not to work, I'm all ears as
> to *why* :)

Yup, that's a very ugly one. Given this is an eval box with an option
to buy, I'd rather send it back.

Other then the register function, the box appears to be a very nice
one. Maybe a little pricey, but would bet it fits into a very large
number of businesses/homes very nicely. 

Rich