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

[Dave Wolven]

Your call is entering your dialplan under the context sip.
The demo is probably an include to the default context.

To get the to work as is....you can try the following:

1. create sip context in extensions.conf, and include the other
contexts.
	[sip]
	include=>default
	include=>demo

2. change the context for your ATA-186 by changing
	context=default.

One of these should do the trick...
Dave



On Tue, 2003-05-13 at 02:33, Miguel Cruz wrote:
> 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
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