Miguel Cruz
2003-May-12 23:33 UTC
[Asterisk-Users] Newbie: Getting demo to work via ATA-186
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@10.0.5.209;tag=3400102590
To: <sip:1000@10.0.5.209;user=phone>;tag=as0b10acca
Call-ID: 2304536916@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@10.0.5.209;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.5.208:5060
From: sip:ata1@10.0.5.209;tag=3400102590
To: <sip:1000@10.0.5.209;user=phone>;tag=as0b10acca
Call-ID: 2304536916@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
Dave Wolven
2003-May-13 09:12 UTC
[Asterisk-Users] Newbie: Getting demo to work via ATA-186
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@10.0.5.209;tag=3400102590 > To: <sip:1000@10.0.5.209;user=phone>;tag=as0b10acca > Call-ID: 2304536916@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@10.0.5.209;user=phone SIP/2.0 > Via: SIP/2.0/UDP 10.0.5.208:5060 > From: sip:ata1@10.0.5.209;tag=3400102590 > To: <sip:1000@10.0.5.209;user=phone>;tag=as0b10acca > Call-ID: 2304536916@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 > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >