[asterisk-dev] [Code Review] SIP: Pineapple

Michiel van Baak michiel at vanbaak.info
Thu Oct 22 06:18:41 CDT 2009

On 11:20, Thu 22 Oct 09, Alexander Harrowell wrote:
> On Thursday 22 October 2009 10:47:18 Olle E. Johansson wrote:
> > 22 okt 2009 kl. 11.30 skrev Nick Lewis:
> > > oej
> > >
> > > re new types: I like the proposal to have peer types related to the
> > > actual network architecture rather than the barmy type=user/peer/
> > > friend
> > > but I find the actual words you have chosen to be confusing. The
> > > relationship that you name "service" is what I regard as a sip trunk
> > > and
> > > which I get from my internet telephony service provider. 
> Yes - is it really a common use case to have end points or trunks that are 
> one-way (as the current typing implies)? I'm sure there will be fancy 
> deployments that have phones attached that are only ever used for inbound, or 
> that send their outbound traffic to a different carrier than they receive inbound 
> from. But I would suspect 90-odd % of Asterisk instances have a SIP carrier on 
> one side carrying both inbound and outbound (i.e. a "friend" in currentspeak) 
> and Linksys desk phones on the other that both receive and place calls. 

It's pretty common to have a couple of ITSP's configured to do
least-cost-routing based on the number you are calling.
One of the ITSP's will be responsible for inbound calls on your main
But in mosts setups we did we have more of those because of DID's in
different countries etc.

And we have a good couple of setups that have an asterisk box specific
for routing, it grabs the calls from ITSP and landlines and routes those
calls to other boxen. Most of them use a different route when setting up
an outbound call.

Many many possibilities that dont match the simple setup you described.

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Michiel van Baak
michiel at vanbaak.eu
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