Hopefully this will save time for someone else searching for help with this. This configuration assumes that you will not be using the router functionality of the HT-503. It also assumes that you're setting this up for connection to a PSTN trunk in the US. If you are in another country, you will need to make changes to various options including Caller ID settings.
First prepare the following information. You will use it in various steps along the way.
[Asterisk_IP] - IP address of your Asterisk box.
[HT503_IP] - Static IP address you will assign to your Grandstream HT-503.
[PSTN_Number] - The phone number of the incoming line you will be using.
[Trunk_Number] - An arbitrary number you will assign to the trunk (for example, 9595).
[Password] - The password you will assign to the trunk connection.
In each step, replace the entire italicized portion (including the brackets) with the respective information:
Create a new SIP trunk
Create a new SIP trunk and populate the entry with the following details. Any details not specified here should be left blank.
Maximum Channels: 1
Trunk Name: [Trunk_Number]
PEER Details:
context=from-trunk
dtmfmode=rfc2833
host=[HT503_IP]
secret=[Password]
type=friend
port=5062
Make sure "USER Context", "USER Details" and "Register String" are all blank.
Click "Submit Changes"
Create a new Inbound Route
Create a new Inbound Route and populate the entry with the following details. Any details not specified here should be left blank.
Description: HT503
DID Number: [PSTN_Number]
Under "Set Destination" you should select the extension or ring group you want to send incoming calls to.
Click "Submit Changes"
Assign Trunk to Outbound Route
Select or create an Outbound Route and add this trunk to the "Trunk Sequence".
Click "Submit Changes"
Click "Apply Configuration Changes"
Click "Continue with reload"
Configure HT-503 BASIC SETTINGS
On the HT-503 under the "BASIC SETTINGS" tab, configure the following options. Any details not specified here should be left at their default settings unless you know that your specific configuration requires different options.
IP Address - Select "statically configured as" and enter the [HT503_IP] you specified earlier.
Device Mode - Select "Bridge".
Enable LAN DHCP - Select "No".
User ID - enter the [PSTN_Number] you specified earlier.
Click "Update"
Configure HT-503 FXO PORT
On the HT-503 under the "FXO PORT" tab, configure the following options. Any details not specified here should be left at their default settings unless you know that your specific configuration requires different options.
Primary SIP Server: [Asterisk_IP]
SIP User ID: [Trunk_Number]
SIP Registration: Select "No".
Number of Rings: 2
(Number of rings for a PSTN incoming call before FXO port answers to accept VoIP number)
PSTN Ring Thru FXS: Select "No".
Wait for Dial-Tone: Select "No".
Stage Method: 1
Click "Update"
Click "Reboot"
That should do it. Just plug the "Line" port of the HT-503 into your PSTN jack and you will be able to place/receive calls.
A few notes:
1) When you make an outbound call, you will hear one ring before the call is actually placed on the trunk. This can seem odd when the number you're calling is busy because you'll hear it ring once and then you'll hear a busy signal. I'm still new to this so there may be an option to eliminate this behavior. I'm sure someone will chime in with details on that.
2) Incoming calls will not start ringing on your VoIP phones immediately. It takes two rings before the HT-503 passes the call to Asterisk. You can shorten this to 1 ring but then you will lose incoming Caller ID.
3) Let me repeat... I am new to this. These steps worked to get things set up on my system. This may not be the "best" way to do it but it's at least a starting point. Please feel free to post corrections, clarifications or suggestions for improvement.
mS
Member Since:
2009-06-23