Jonathan! You don't know how much that simple explanation has helped me understand Asterisk. Well done. Well said. And to the point clearly.

I would hope this could find it's way onto the Asterisk Wiki and be the *first* thing someone reads when looking at the documentation about sip.

Thanks a million!

J.

Johnathan Corgan wrote:

Jeffrey Starin wrote:

I have a little confusion about the general settings (other than the register values) in the SIP
General area.


[snip]

However, I'm confused as to the purpose of the
"general" settings -- to what or which connection do they apply? Since
the context suggested for the general settings is something like
"nothing" to avoid unwanted sip calls, I'm confused as to the purposse
of values in the general section since all parameters for communications with VOIP providers is contained in the contexts such as I specified above, i.e., [FWD] or for example [BROADVOICE]. Can someone shed some light on that for me?


Well, your confusion is understandable, given the way sip.conf works. Parameters which affect incoming calls are not separated from those that affect outgoing calls, so it's easy to get mixed up (well, for me, anyway.)

In the [general] section the parameters become the defaults used unless overridden in a specific peer section. Also, if an incoming or outgoing SIP call doesn't match a specific peer section, these parameters get used.

So, for example, if you don't want any incoming SIP calls that aren't from a known provider, you can set the default context to something innocuous as you describe above and the call will get rejected as a non-existent context. This is what you described in our original mail.

But also in this [general] section are settings for *outbound* calls using SIP that aren't using a specific peer section. This can be done with the Dial command, using a dial string such as IP/[EMAIL PROTECTED] where provider.com is not listed in sip.conf. This might happen, say, in the case of using the ENUM lookup capability, where the outbound SIP address of a phone number is determined dynamically at call time rather than pre-configured in extensions.conf/sip.conf.

Personally, I'd like to see this changed so there are two 'general' sections--one for default parameters to use unless overridden when there *is* a peer section below, and a different one to describe parameters to use when the remote peer is not previously known. I know there are ways to accomplish this with the existing sip.conf structure but it seems very counter-intuitive.

-Johnathan

_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to