asterisk users - Aug 2007 - Several doubts on Asterisk as an UAC

Hi,

I'm new to Asterisk and I've been trying to configure it to talk to
several SIP providers (such as FWD). I found that, although there are
some "recipes" on how to do it, there are few documents that really
explain *why* the settings are used, and overall I found very little
documentation on sip.conf.

I've been using this page as a reference:
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
It's very good. However, there's no explanation on "register"
command,
for instance. There are also some other things that are not completely
clear.

That's why I wanted to ask lots of questions about it, and hopefully
I'll get some good answers here.

I'll focus only on having Asterisk being a SIP *client* (I believe this
is called UAC [User Agent Client] in the SIP world, right?) connecting
to a SIP provider (such as FWD). I'm using asterisk 1.4.6, so I would
like to talk *only* about configuration on asterisk 1.4.x.



* SIP channels on outgoing calls

If on my sip.conf I have a section [myprovider], it always creates a new
SIP channel "SIP/myprovider", right? If I want to use it on
extensions.conf to call extension 464646 there, I can use:
	Dial(SIP/myprovider/464646)
or:
	Dial(SIP/464646 at myprovider)
Is that right?


If I don't want to keep the section on sip.conf, and myprovider's host
is "hostname.myprovider.com", and I login with user "myuser"
and secret
"mypasswd", I can also use this information directly on the
"Dial" like
this:
	Dial(SIP/myuser:mypasswd at hostname.myprovider.com/464646)
Is that right?


Do I always have to authenticate to make outgoing calls? For instance,
could I do just this?
	Dial(SIP/hostname.myprovider.com/464646)
Would it work for some outgoing numbers but not others? I believe with a
provider that allows me to do outgoing PSTN calls it wouldn't work, but
with free providers it might... someone has some info on this?



* Difference between type=user and type=peer

I saw this:
http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer

But I still didn't understand completely how it works...

First of all, the configuration type=... will affect only *incoming*
calls, right?

If I got it right, when Asterisk receives a call it will try to match to
all the type=peer sections by matching the "Host:" field of the INVITE
packet to the host=... configurations of type=peer sections. Right?

If that fails, it will ask for an authenticated INVITE, then it will
take the username/secret from the authenticated INVITE packet, and then
will look for type=user sections and match with username=... fields.

Now, what does happen for type=friend sections? Can they have both
host=... and username=... fields and they'll match on both?

Could someone really explain what is the algorithm (or at least the
idea) of how the call is matched when it's being received?



* context= in [default] and on outgoing calls

Ok, so if I have [myprovider] section with context=something. When I do
an outgoing call by using "Dial(SIP/myprovider/464646)", does
context=... affect anything?

As I understand it, it only affects incoming calls, but I might be
wrong.

Another thing, the setting of context=... on [default] section will
affect all [provider] sections without context=..., right? What if I
don't specify any context on [default], what would be the default
context? What if there's no context or an invalid context on a section,
what would happen to incoming calls that match that section?



* What exactly are "fromuser" and "fromdomain"?

As I get it, they're used only in outgoing calls, right?

As I understand, they change some fields on the INVITE packet sent to
the provider. Right?

Now, why do I need them?

If I don't specify them, what is the behaviour? What's the default value
for these options?



* The use of "register => ..."

When I use "register", the *only* thing it does is that Asterisk will
send periodic REGISTER packets to the provider, right? This is useful
for the provider to know to which IP it should direct calls that it
receives for my extension.

When I use "register => ...", does it create a "SIP/..."
channel? I
think it doesn't, but if you use the same syntax, which is
"SIP/myuser:mypasswd at hostname.myprovider.com/${EXTEN}" it will
create a
channel "on-the-fly", am I right?

When I use "register => ...", if after that I use
"SIP/hostname.myprovider.com/${EXTEN}", will it use the same
username/secret I used for registering?

Is it possible to register without a secret? Does it make sense?

Do I always have to put "register => ..." commands on the [default]
section? Does it make sense to put them on other sections? What would be
the side effects of that?



* The use of "insecure"

If I got this right, I should use "insecure" when I want specific
tasks
to be done without authentication.

For example, if I'm registered to a provider and someone wants to call
me, the provider's host will send me an INVITE packet, right?

If I don't use "insecure", the Asterisk will answer with an error
code
that requires the provider to authenticate. If the provider can
authenticate *with the same secret I use to register and to authenticate
to the provider*, then it will authenticate the INVITE and the call will
complete.

But I believe this is not the case for most providers, usually they
don't authenticate calls to UAC's, am I right?

In this case, I should use "insecure=invite" for these providers.

This has a security issue involved, that is, anyone on the Internet
might send me an INVITE packet with "Host: hostname.myprovider.com" on
the SIP header, and my Asterisk box would accept the call as it was
really coming from myprovider, that means, it will route the call to the
context=... set on the [myprovider] section where there's the definition
of "host=hostname.myprovider.com", right?



* "canreinvite=no"

I believe is different if I use "canreinvite=no" for my SIP phone and
"canreinvite=no" to a provider.

Using "canreinvite=no" means that for every call I make using my SIP
phone, throught the Asterisk box, to a SIP provider, Asterisk will *NOT*
try to send a redirect to the SIP phone saying that the phone should
send RTP to the provider's servers or even to the other endpoint
directly, instead of sending RTP to the Asterisk box and then the
Asterisk box sending it to the endpoint.

Using "canreinvite=no" on a provider configuration, for every call
using
a SIP provider, I'll force Asterisk to send all audio data to the
provider and won't allow Asterisk to send RTP directly to the other
endpoint.

Am I right?

In this case, the recommended setting would be using "canreinvite=no"
for my SIP phones on my internal network, because I want them to talk to
the Asterisk server only. But, on the other hand, I should probably set
"canreinvite=yes" on the provider configuration, because then I may
end
up having a more direct path to the destination, without using the
provider as a proxy, and also with less latency. Am I right?



Well, I think that's it... I thank you very much for having the time to
see my doubts!

Thanks for making Asterisk the great product it is.

Filipe