[Asterisk-Users] SER, Asterisk, SIP proxy, routing, redirection - confused

[Mike Hansford]

I am fairly new to Asterisk / VOIP and have been playing around with it for
long enough to have a whole lot of questions so far without answers.

Presently I’m running Asterisk (v.1.0.7) on a Debian Sarge installation with
2 soft phones (for testing purposes). A live deployment will probably have a
dozen-odd extensions. I wish to have both SIP and PSTN services exposed to
the outside and will probably install an appropriate Digium card to allow me
to connect PSTN lines. We pay ransom to Microsloth for our company network.

I am reading that Asterisk does not provide SIP proxying services however
proxy services are “very important” (one reference said “critical”) to
routing in SIP as it provides for dynamic rewriting, redirection and
inter-domain routing. In Asterisk, how are these functions meant to work? As
far as I can tell, it cannot perform inter-domain routing as it has no
proxying capability but apparently provides redirection and rewriting
services. Am I going to require the services of SER (perhaps in a gateway
role) in order to achieve any or all of these functions or will Asterisk
alone provide it? I have been reading the SER documentation and it seems to
be very capable however I think that establishing the dial plan and
voicemail in Asterisk may be a simpler and clearer process. So my next
question may be how are people deploying Asterisk with a separate proxy
server? Early on I was reading that a proxy is mainly useful in a large
environment (thousands of extensions) in order to reduce the load on the
Asterisk server however this doesn’t seem to mesh with what I’m reading now
about a proxy providing SIP routing services.

To date, I have only been able to set up Asterisk with fixed extension
numbers with no facility for authenticating a particular user at a terminal.
Being able to tell Asterisk where a particular user is and direct calls to
them is one of the core capabilities of SIP and is one of the key reasons
why we want to deploy it into our office. Yet I’ve seen no documentation on
how to do this.

As you can probably gather, I’m rather confused about how to develop /
deploy a VOIP solution. There is much written about the topic however they
seem to say conflicting things


Any help would be appreciated.
Mike
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