The Grandstream HandyTone HT-488 is a low-cost SIP analog adapter that offers one FXS port and one FXO port to connect one analog telephone and one POTS line to the unit.

The scenario

We want to install the unit with an Asterisk server whick IP is 10.10.3.5 and want to register the telephone attached to the unit as SIP/32, while the FXO line will be seen by Asterisk as SIP/33.

Installing the HT-488

As it ships, access for the web interface is disabled for users accessing from the WAN. To enable it, connect the LAN port of the unit to a PC with a cross-cable patch, use the unit’s own DHCP server to assign address to the PC and logon by navigating to http://192.168.2.1 (password admin ) - go to Basic settings and set WAN side http access. Save and reboot.

Connect the unit to your LAN through the WAN port and connect to it through its DHCP-assigned address (to see what that is, connect a phone to the FXS port and digit **02** to have the unit read out the assigned IP address). If the unit did not get an IP address, select the option **01** and then 9 to switch from fixed-ip to DHCP mode.

Firmware upgrade

It is VERY IMPORTANT that you set the unit with a modern firmware, or you’ll have a number of problems. Luckily this is very easy: check the current TFTP server address on Grandsteram site (see below) and enter it under the configuration page, and set “Yes, check for upgrade every 1 minute”. Reboot. The unit light should go red while the unit downloads and installs the new firmware, it will take like 5 minutes to update.

Unit configuration

  • Basic settings Under NAT/DHCP Server Information & Configuration

    WAN side http access: yes Reply to ICMP on WAN port: yes Number of Rings: 0 Forward to VoIP: s FXO One Stage Dialing: yes

The Forward to Voip option must be set to “s” in order to have incoming FXO calls sent over to the Asterisk server.

  • FXS PORT

    SIP Server: 10.10.3.5 Outbound Proxy: 10.10.3.5 SIP User ID: 32 Authenticate ID: 32 Authenticate Password: XXX Name: (leave blank)

    SIP Registration: yes Unregister On Reboot: yes

    Send DTMF: via RTP (RFC2833) Send Flash Event: Yes Enable Call Features: No NAT Traversal (STUN): No No Key Entry Timeout: 4

    Preferred Vocoder: PCMU / PCMA Silence Suppression: No Fax Mode: Pass-through

  • FXO PORT

    SIP Server: 10.10.3.5 Outbound Proxy: 10.10.3.5 SIP User ID: 33 Authenticate ID: 33 Authenticate Password: XXX Name: (leave blank)

    SIP Registration: yes Unregister On Reboot: yes

    Send DTMF: via RTP (RFC2833) Send Flash Event: No NAT Traversal (STUN): No

    Preferred Vocoder: PCMU / PCMA

    Silence Suppression: No Fax-mode: Pass-through

    PSTN AC Termination: 270 Ohm Enable PSTN Disconnect Tone Detection: yes PSTN Silence Timeout: 20 Enable Current Disconnect: Yes

Save and reboot

Asterisk configuration

In SIP.conf enter the following piece of code:

; FXS port
[32]
type=friend
secret=XXX
callerid="My FXS" <32>
host=dynamic
nat=no
canreinvite=no
disallow=all
allow=ulaw
context=sip
qualify=yes
dtmfmode=rfc2833

;  FXO port
[33]
type=friend
secret=XXX
callerid="My FXO" <33>
host=dynamic
nat=no
canreinvite=no
disallow=all
allow=ulaw
context=in-sip
qualify=yes
dtmfmode=rfc2833

Once the phones are registered, you have a standard SIP phone as SIP/32 that will send calls to the context sip, while the SIP/33 POTS line can be dialled as Dial(SIP/33/number,30) and will send incoming calls to s@in-sip.

To check that the lines are working, log in to Asterisk and issue:

pbx*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
33/33                      10.10.3.103      D          5062     OK (5 ms)
32/32                      10.10.3.103      D          5060     OK (5 ms)

See also: