Does anyone know how to setup the HT488 to do call termination to PSTN using my asterisk box?

I want to spread 24 of these HT488 around to have 24 lines to which I will terminate traffic to.

My HT-488 is finally working with Asterisk/trixbox. This is all on a private LAN. (I found info scattered on various sites, and changed a few things) Here is a list of what I did:

1. Upgraded 488 firmware to 1.0.3.96 (used TFTP server at 168.75.215.190)
2. Make changes to the 488 pages via web browser
2.1. BASIC section, set:
2.1.1. Number of Rings: 0
2.1.2. FXO One Stage Dialing: Yes
2.2. FXS section, set:
2.2.1. SIP Server: IP of trixbox
2.2.2. Outbound Proxy: IP of trixbox
2.2.3. SIP user ID: 432
2.2.4. Authenticate ID: 432
2.2.5. Password: password
2.2.6. SIP Registration: yes
2.2.7. Unregister On Reboot: yes
2.2.8. Send DTMF: via RTP (RFC2833)
2.2.9. Send Flash Event: yes
2.2.10. Enable Call Features: no
2.2.11. NAT Traversal (STUN): no
2.2.12. No Key Entry Timeout: 4
2.2.13. Preferred Vocoder: PCMU, PCMA
2.2.14. Silence Suppression: no
2.2.15. Fax Mode: Pass-Through
2.3. FXO section, set:
2.3.1. SIP Server: IP of trixbox
2.3.2. Outbound Proxy: IP of trixbox
2.3.3. SIP user ID: 433
2.3.4. Authenticate ID: 433
2.3.5. Password: password
2.3.6. SIP Registration: yes
2.3.7. Unregister On Reboot: yes
2.3.8. Send DTMF: via RTP (RFC2833)
2.3.9. Send Flash Event: no
2.3.10. NAT Traversal (STUN): no
2.3.11. Preferred Vocoder: PCMU, PCMA
2.3.12. Silence Suppression: no
2.3.13. Fax Mode: Pass-Through
2.3.14. PSTN AC Termination: 270 ohm
2.3.15. Enable PSTN Disconnect Tone Detection: yes
2.3.16. PSTN Silence Timeout: 20
2.3.17. Enable Current Disconnect: yes
3. Within trixbox, set up the following:
3.1. Create new extension (generic SIP Device)
3.1.1. Extension: 432
3.1.2. Name: 488FXS
3.1.3. secret: password
3.1.4. dtmfmode: rfc2833
3.1.5. canreinvite: no
3.1.6. context: from-internal
3.1.7. host: dynamic
3.1.8. type: friend
3.1.9. nat: no
3.1.10. qualify: yes
3.2. Create new extension (generic SIP Device)
3.2.1. Extension: 433
3.2.2. Name: 488FXO
3.2.3. secret: password
3.2.4. dtmfmode: rfc2833
3.2.5. canreinvite: no
3.2.6. context: from-pstn
3.2.7. host: dynamic
3.2.8. type: friend
3.2.9. nat: no
3.2.10. qualify: yes
3.3. Add Custom Trunk
3.3.1. Maximum channels: 1
3.3.2. Custom Dial String: SIP/,w$OUTNUM$@433
3.4. Add Outbound Route
3.4.1. Route Name: HT488-OUT
3.4.2. Dial Patterns: . (a period)
3.4.3. Trunk Sequence: SIP/,w$OUTNUM$@433
3.5. Add Incoming Route
3.5.1. Description: IncomingPSTN
3.5.2. Set Destination: Extension: softphone on PC

I think that covers all the little steps. In the panel screen, the FXO and FXS ports show up as extensions, not in the trunk section.

So it works now... but there are two things that I can't get working, and from all the searching I've done, I don't think there is a work around for them.

The HT-488 does not pass callerid to Asterisk, making this ATA useless for what I intended.

When I try to dial out through the FXO using the analog phone connected to the FXS port, I get the busy signal and get the following error: (it will work if I dial *00 first to bypass VoIP altogether)

   — SIP/433-09c09068 is ringing
   — SIP/433-09c09068 answered SIP/432-09bccea8
   — Packet2Packet bridging SIP/432-09bccea8 and SIP/433-09c09068
 == Spawn extension (macro-dialout-trunk, s, 32) exited non-zero on 'SIP/432-09bccea8' in macro 'dialout-trunk'
 == Spawn extension (macro-dialout-trunk, s, 32) exited non-zero on 'SIP/

I have 2 ht-488 1.0.3.96

I have truble getting line

i set 2 fxs for phones 200 + 201

I set 2 ext 250 + 251

I set 2 trunk 250 +251


If Trunk Sequence 0 is 250 and Trunk Sequence 1 is 251 then I can get an out line on phone 201 on line 200 I get busy sign

If Trunk Sequence 0 is 251 and Trunk Sequence 1 is 250 then I can get an out line on phone 200 on line 201 I get busy sign

If I set only one 488 I have no out line - I get busy sign

Please help !!!

As of Firmware 1.0.3.86 (I am running 1.0.3.96) the dialing issues are fixed!

On the Basic Settings configuration page for the HT-488 is an option for FXO One Stage Dialing

Set this to YES.

You can now change your dialplans to something much more sane:

pstn-calls-local
exten => _XXXXXXX,1,Dial(SIP/pstn/${EXTEN})



Happy Dialing!

For those that have been having problems setting up the unit, here is a copy of my sip.conf entry


    pstn
    host=dynamic
    type=friend
    context=from-pstn
    canreinvite=no
    dtmfmode=rfc2833
    secret=pstn



How to get Incoming to work
1. On the Basic Settings page set Number of Rings to 0
2. Set Forward to Sip to the letter s

In your extensions.conf file
from-pstn
exten => s,1,Dial(${PHONES},10)
exten => s,2,Hangup()

Note: Your milage may vary

Can we merge this page with Grandstream Handytone 488?

I bought the HT 488 back at christmas and spent a while trying to get it to work (as documented on this page) using the two different trunks method. Never could get the FX0 port working. Eventually I gave up and the box went in the cupboard and has never been used.

Revisited this page last week after upgrading my asterisk box to Trixbox 2.2 (and Freepbx 2.2.2) and noticed details of firmware update and Joeseph's FreePBX config which looked a lot simpler. Updated to latest firmware. So the FXS port works fine and can dial out over VOIP provider from telephone handset, but STILL a busy tone with the FXO. Can also bypass asterisk with the default '00' and dial out over PSTN fine.

My setup is as follow... and help MUCH appreciated:

1 extension (number '100') for FXS - working fine

As per Joseph instruction, extra extension for FXO connection - extension '101' (username, pass '101', port to 5062)

Custom Trunk: SIP/,,,w$OUTNUM$@101
Max Channels = 1
Dial Rules = .
Cust Dial String = SIP/,,,w$OUTNUM$@101

Outbound route: HT488-OUT
Dial patterns = .
TRUNK sequence = SIP/,,,w$OUTNUM$@101

In the FXO config settings username pass etc set to '101'

In the freepbx GUI panel both the 100 and 101 extensions are lit up green, my custom trunk is not visible in this panel (my VOIP trunk is)., although both are visible in the freepbx setup (and have applied configs). I have asterisk logging set on full and can provide if helpful.

Please please someone help before I throw this little black box with its irritating flashing green light out the window!!