[Asterisk-Users] Questions from an Asterisk newbie

[ty.roach at acecomm.com]

I have just installed asterisk in the hopes of operating a very simple VoIP
demo.  The demo environment is as follows:

Asterisk 1.0.2 installed on a Fedora 2 Linux laptop.  The laptop is
connected to a hub along with two Cisco 7960 IP phones (SIP enabled).  I've
manually configured the phones setting the IP address of the phones, phone
names (extensions), the IP address of the SIP proxy (Asterisk server?).

I have not made any modifications to any of the asterisk configuration
files.

I run asterisk ('asterisk -cv') from the command line just to see what
happens.  Essentially, I get messages from both SIP phones indicating that
registration is failing (I guess not such as surprise since I haven't
configured anything).

For starters, I was hoping that some of the experts on this board could
give me some tips on what I need to do to allow one phone to successfully
call the other phone.  I did a similar thing several years ago using a SIP
proxy server (from Dynamicsoft, albeit, with help from their support
group).

Any advise would be greatly appreciated.  Thanks and advance.

Ty

P.S.  I've included command line output from my asterisk console below...


*CLI> sip debug
SIP Debugging Enabled
*CLI>
*CLI>
*CLI>
*CLI>

Sip read:
REGISTER sip:172.20.23.201 SIP/2.0
Via: SIP/2.0/UDP 172.20.23.211:5060
From: sip:4444 at 172.20.23.201
To: sip:4444 at 172.20.23.201
Call-ID: ce30300-411dcd5-8f0953-2e323731 at 172.20.23.211
CSeq: 101 REGISTER
Contact: <sip:4444 at 172.20.23.211:5060>
Expires: 3600
Content-Length: 0


9 headers, 0 lines
Using latest request as basis request
Sending to 172.20.23.211 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.20.23.211:5060
From: sip:4444 at 172.20.23.201
To: sip:4444 at 172.20.23.201;tag=as106566ef
Call-ID: ce30300-411dcd5-8f0953-2e323731 at 172.20.23.211
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4444 at 172.20.23.201>
Content-Length: 0


 to 172.20.23.211:5060
Nov  5 13:37:01 NOTICE[-159417424]: chan_sip.c:7571 handle_request:
Registration from 'sip:4444 at 172.20.23.201' failed for '172.20.23.211'
Scheduling destruction of call
'ce30300-411dcd5-8f0953-2e323731 at 172.20.23.211' in 15000 ms
Destroying call 'ce30300-411dcd5-8f0953-2e323731 at 172.20.23.211'


Sip read:
REGISTER sip:172.20.23.201 SIP/2.0
Via: SIP/2.0/UDP 172.20.23.212:5060
From: sip:3005 at 172.20.23.201
To: sip:3005 at 172.20.23.201
Call-ID: 2ae30300-4302418-8f1c2b-2e323731 at 172.20.23.212
Date: Fri, 05 Nov 2004 18:38:54 GMT
CSeq: 101 REGISTER
Contact: <sip:3005 at 172.20.23.212:5060>
Expires: 3600
Content-Length: 0


10 headers, 0 lines
Using latest request as basis request
Sending to 172.20.23.212 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 172.20.23.212:5060
From: sip:3005 at 172.20.23.201
To: sip:3005 at 172.20.23.201;tag=as66d562fd
Call-ID: 2ae30300-4302418-8f1c2b-2e323731 at 172.20.23.212
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3005 at 172.20.23.201>
Content-Length: 0


 to 172.20.23.212:5060
Nov  5 13:37:36 NOTICE[-159417424]: chan_sip.c:7571 handle_request:
Registration from 'sip:3005 at 172.20.23.201' failed for '172.20.23.212'
Scheduling destruction of call
'2ae30300-4302418-8f1c2b-2e323731 at 172.20.23.212' in 15000 ms