bubblestar

If I don't get you wrong, you have to put your desired asterisk IVR extension into one of the eight Dial Plans in the DIAL PLAN section under PSTN TAB.  Once you are accepted with the correct PIN, you will be put through to the IVR extension as below:

For example:

6001@your_asterisk_server_IP_address

bubblestar

Glad that you did it.

Actually, the principle of accessing the extension not depends on real extension you created.  It depends on the extension@your_server_ip that matches the incoming registration request.  Hence, a virtual extension also works.  This method is always used to apply in URI dialing
We don't need a real extension in users.conf.

For example, you can create a virtual extention called HKSAR and put it into your extensions.conf like:

[incoming]
exten => HKSAR,1,Dial(SIP/6001,,r)

When someone calls HKSAR@your_server_ip, it will be directed to your * Server and if it matches with the one in incoming context, your real extension 6001 will ring.

The concept here is very useful in terms of extensions management and security.  You can change the named extension without changing your real extension number.  Also, you don't need to disclose your real extension to strangers to strengthen security.  In addition, it will be more user-friendly for your friends to remember your NAME rather than number.  What do you think if you use ckleea@ckleea.com.hk?  If you do, I can throw away my phone book right away

bubblestar

Try this

[incoming]
exten => HKSAR,1,Dial(SIP/6001,,r)
exten => h,1,Hangup()                     ; signal to force hang up

bubblestar

Yes.  SPA3000 has such silence detection feature as well.  It depends on which ways suit you best.  I use both ways in SPA3102 and Asterisk Server extension.conf