I'm using a DTA310 from www.Packet8.net. This is a 1 port FXS
SIP device. It's about US$90 is you sign up for the packet8
service and about $130 if you don't. It has 1xPower,
1x100BaseT, 1xRJ-45 and a 1xRJ-11 ports. It has LEDs for power,
link, phone in use, and message waiting. It is configured via a
built-in web page. I have the DTA310 configured to talk to my
Asterisk server rather than the Packet8 service.
This morning when I tried to use the analog phone connected to
the DTA310 I didn't get a dialtone (which indicates it is not
talking to a SIP server). I power cycled the unit and it came
back fine. (last night was the first time I had the device
powered on overnight) When I have the device powered on I get
the following debug stuff every little while. Notice near the
end Asterisk is complaining about getting a response about a
call it doesn't know about.
SIP Debugging Enabled
*CLI> DEBUG[2051]: File chan_sip.c, Line 401 (create_addr): Setting NAT on
RTP to 0
Interface is eth0
IP Address is 172.16.17.7
11 headers, 1 lines
XXX Need to handle Retransmitting XXX:
NOTIFY sip:2111@172.16.17.51 SIP/2.0
Via: SIP/2.0/UDP 172.16.17.7:5060;branch=4b0dd9a4
From: "asterisk" <sip:asterisk@172.16.17.7>;tag=4083ff58
Contact: <sip:asterisk@172.16.17.7>
To: <sip:2111@172.16.17.51>
Call-ID: 70067c4519aa10d161636c627959a811@172.16.17.7
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: text/plain
Content-Length: 20
Message-Waiting: no
(no NAT) to 172.16.17.51:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.17.7:5060;branch=4b0dd9a4
From: asterisk<sip:asterisk@172.16.17.7>;tag=4083ff58
To: sip:2111@172.16.17.51
Call-ID: 70067c4519aa10d161636c627959a811@172.16.17.7
CSeq: 102 NOTIFY
Server: DTA SIP/0.11.7 NNOS/VR30
Content-Length: 0
8 headers, 0 lines
Interface is eth0
IP Address is 172.16.17.7
DEBUG[2051]: File chan_sip.c, Line 3431 (handle_request): That's odd... Got
a response on a call we dont know about.