Trixbox and Grandstream ATA-488
Submitted by dbuccheri on Fri, 07/07/2006 - 3:04pm.
Hello,
Can anybody give me a step by step instruction on using a grandstream ata488's FXO port to connect a PSTN line to Trixbox as a trunk. If you could walk me though the FXO screen in the ATA-Web configuration and the Trunk / Sip Screen of trixbox, that would be great.
Thanks in advance for any help.
 
Re: Trixbox and Grandstream ATA-488
Thu, 07/06/2006 - 6:43pm
Good luck.... I bought 2 when voip supply was out of spa-3000's
I could not find anything on setting these up for FXO. I ended up RMAing after spending way to long trying to get it going.
 
Re: Trixbox and Grandstream ATA-488
Fri, 07/07/2006 - 3:04pm
I have a working HT-488 configured as of the instructions found in:
http://www.voip-info.org/wiki/view/HT-488
search for Mervyn Sirju comment posted Friday 05 of May, 2006
I had to do everything that he suggested, and it worked! Please be aware that there is a new Beta firmware for this device, that for me solved a nasty echo I had with the FXO port.
I reproduce Mervyn's post here for completion:
After much struggling, I've managed to get the FXO port on the HT-488 to work perfectly as a trunk with FreePBX (AMP/Asterisk@Home)
You'll need to set up two trunks (one as a SIP trunk and one as a custom trunk) and you might need to make some modifications to your extensions.conf (this is to correct a bug in FreePBX).
Step 1:
Add a SIP trunk with these details:
Maximum Channels: 1
Trunk Name: HT-488
PEER Details:
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
host=dynamic
secret=
Leave "USER Context", "USER Details", and "Register String" all blank
Step 2:
Add a CUSTOM Trunk with these details:
Maximum Channels: 1
Custom Dial String: SIP/HT-488,,D(w$OUTNUM$)
Step 3:
Modify your extensions.conf as follows:
In the macro-dialout-trunk section, look for the following lines:
exten => s,21,Dial(${pre_num:4}${the_num}${post_num})
exten => s,22,Goto(s-${DIALSTATUS},1)
Replace it with the following lines:
exten => s,21,Set(full_dialstring=${pre_num:4}${the_num}${post_num})
exten => s,22,Set(DialArg1=${CUT(full_dialstring,\,,1)})
exten => s,23,Set(DialArg2=${CUT(full_dialstring,\,,2)})
exten => s,24,Set(DialArg3=${CUT(full_dialstring,\,,3)})
exten => s,25,Dial(${DialArg1},${DialArg2},${DialArg3})
exten => s,26,Goto(s-${DIALSTATUS},1)
Step 4:
In your HT-488's web based configuration, go to the 'BASIC SETTINGS' Section
Set number of rings to answer PSTN line from '4' to '1'
Set "Forward to VoIP" to : s
Step 5:
In your HT-488's web based configuration, go to the 'FXO Port' Section
Enter the login details that you specified in 'PEER Details' in Step 1
Be sure to set your User ID and Auth ID to be 'HT-488'
Step 6:
Use: "SIP/HT-488,,D(w$OUTNUM$)" as your trunk when setting up your outgoing routes.
That should be it, if it doesn't work for you, you may need to tweak it a bit, email me at mervstar@hotmail.com if you need help with the tweaks.
 
Member Since:
2006-07-02