N.B. – As of 2005/09/29 it appears this device is no longer on the D-Link website.
A small NAT router with built-in VOIP SIP ports and a QoS mechanism
It has 2 FXS ports and 1 FXO port.
You can configure a table to route calls by dialed digits to the FXO port (connected to a PSTN line) or a SIP connection.
The 1120S (SIP firmware) device works great with Asterisk. There have been mixed results with the MGCP version (1120M). It is possible to convert an 1120M to an 1120S by reflashing the firmware, but redistributing the firmware may not be legal.
Provider Specific settings (for DLink DVG-1120M)
Some providers have custom firmware that causes the gateway to loose all it’s settings when the device reboots, making it really difficult to get the gateway to work with an asterisk server, not on the local subnet. To resolve this issue:
1. telnet 192.168.15.1 (or whatever the ip is)
2. Login
3. autocfg func 2
4. commit
Now you should be able to play with the settings without any troubles of losing your settings…
User’s Guide
A complete and very useful version of the user’s guide is available at DVG-1120S User’s Guide. This word code explains most features of the unit, whereas, the other PDF version from the D-Link site says next to nothing. This manual is a marked draft but contains the usage of the console port and other features.
Asterisk mgcp.conf settings (1120M)
[000abcdef12] ; device's mac address host = dynamic context = from inside canreinvite = no mailbox=14165551213@default nat=yes threewaycalling = yes transfer = no callwaiting = yes callerid="Line 2" <14165551213> line => aaln/2 context = from-inside mailbox=14165551212@default threewaycalling = yes transfer = no callwaiting = yes nat=yes callerid="Line 1" <14165551212> line => aaln/1 line => *
Notes about the DVG-1120m and asterisk:
-For the hassle to get this device to work with asterisk, it is more convenient to go and buy a SIP adapter for 60$.
-Call waiting doesn’t always work (sometimes it beeps but you can’t pick it up)
-The Primus Canada branded DVG-1120m has several distinct ring tones available (try _ALERT_INFO=1) (There seems to be 4 ring tones. 1. Standard ringtone, 2. two medium length tones. 3. 2 short rings and a third slightly longer ring. 4. one short one long and one short.)
Asterisk sip.conf Settings
[1020] username=1020 type=friend secret=SECRET qualify=no port=5061 nat=never host=dynamic dtmfmode=inband context=internal canreinvite=no callerid="1020"<1020>
Note: The CID name inside quotes on “caller id” must correspond to the “display name” field on the VoIP router’s page below.