[Asterisk-Users] Inbound SIP to demo context

[Steve Blair]

  I have the latest Asterisk installed however the analog
line card I was using doesn't work in the new server, that is ok.
Instead what I want to do is allocate one of my numbers to be the
lead number for the Asterisk system. User who call this will have
their call delivered to our site specific version of the Asterisk
mainmenu via an IP connection.

  I already have SIP connections coming into Asterisk from my
SER box for the purpose of voicemail. This is working fine.
What I cannot get working is for calls from a specific extension
to goto mainmenu.

  I thought I could just define a second context in sip.conf to point
to the mainmenu context in extensions.conf but this isn't working.
Calls just get dropped. Does anyone know or have ideas about
how to get this working?

Thanks