[Asterisk-Users] Newbie: Getting demo to work via ATA-186

[Miguel Cruz]

I've installed Asterisk and configured an ATA-186 as described at this 
link:

   http://www.djernes.org/~shawn/ata186.htm

Unfortunately this guide abruptly ends before it explains how to deal with 
the sip.conf and extensions.conf files.

So I left extensions.conf alone and my sip.conf looks like this:

   [general]
   port = 5060                     ; Port to bind to
   bindaddr = 0.0.0.0              ; Address to bind to
   context = default               ; Default for incoming calls

   [ata1]
   type=friend
   host=dynamic
   dtmfmode=rfc2833
   context=sip
   username=ata1
   secret=ata1

Now I'm stuck.

Whatever I dial on the SIP phone gets me a fast busy. I assumed that since 
I'm using the extensions.conf file as distributed, I could monkey with the 
demo - perhaps dialing 1234 or 1000 might do something. Nope.

Here's the last (and I think relevant) bit of what I see with sip debug on
the console (10.0.5.208 is the ATA-186 and 10.0.5.209 is the Asterisk
box):

   v=0
   o=ata1 236029 236029 IN IP4 10.0.5.208
   s=ATA186 Call
   c=IN IP4 10.0.5.208
   t=0 0
   m=audio 16384 RTP/AVP 0 18 8 101
   a=rtpmap:0 PCMU/8000/1
   a=rtpmap:18 G729/8000/1
   a=rtpmap:8 PCMA/8000/1
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15

   12 headers, 11 lines
   Using latest request as basis request
   Sending to 10.0.5.208 : 5060 (non-NAT)
   Capabilities: us - 14, them - 268, combined - 12
   Non-codec capabilities: us - 1, them - 1, combined - 1
   Looking for 1000 in sip
   Transmitting (no NAT):
   SIP/2.0 404 Not Found
   Via: SIP/2.0/UDP 10.0.5.208:5060
   From: sip:ata1 at 10.0.5.209;tag=3400102590
   To: <sip:1000 at 10.0.5.209;user=phone>;tag=as0b10acca
   Call-ID: 2304536916 at 10.0.5.208
   CSeq: 2 INVITE
   User-Agent: Asterisk PBX
   Contact: <sip:@10.0.5.209>
   Content-Length: 0


    to 10.0.5.208:5060
   Sip read: CLI> 
   ACK sip:1000 at 10.0.5.209;user=phone SIP/2.0
   Via: SIP/2.0/UDP 10.0.5.208:5060
   From: sip:ata1 at 10.0.5.209;tag=3400102590
   To:  <sip:1000 at 10.0.5.209;user=phone>;tag=as0b10acca
   Call-ID: 2304536916 at 10.0.5.208
   CSeq: 2 ACK
   User-Agent: Cisco ATA  v2.15 ata186 (020918a)
   Content-Length: 0

So I guess my questions are:

1) Have I set things up reasonably?

2) If so, am I correct in thinking the demo should work?

3) If so, how would I verify that it does? (or really make anything at all 
happen)

Thanks very much for any advice. I'm stumped.

miguel