01 Dec Grandstream HT503 FXO Trunk
Even though we have used it in the past, we would not recommend it for professional use.
However, please try the below configuration (update the bold values):
1. Asterisk Side
SIP TRUNK
Trunk Description: PSTN
Outbound Caller ID: Your Phone Number
Maximum Channels: 1
Outbound Dial Prefix: 9 (Leave it blank if the call gets disconnected)
Trunk Name: 300
SIP TRUNK: PEER DETAILS
disallow=all
allow=ulaw&alaw
dtmfmode=rfc2833
host=dynamic
context=from-trunk
qualify=yes
nat=yes
port=5062
secret=YOUR PASSWORD
type=friend
username=300
sendrpid=yes
Outbound Rules
Dial Patern: 9|.
2. HT503 Side
Basic Settings
PSTN Access Code: 9
Unconditional Call Forward to VOIP: your_extension@AsteriskIP:5060
Advanced
Call Progress Tones
Dial Tone: f1=425@-13,f2=425@-19,c=200/300-700/800;
Ringback Tone: f1=425@-13,f2=425@-19,c=150/150;
Busy Tone: f1=425@-13,f2=425@-19,c=300/300;
Reorder Tone: f1=425@-13,f2=425@-19,c=200/300-700/800;
Confirmation Tone: f1=425@-13,f2=425@-19,c=200/300-700/800;
Call Waiting Tone: f1=425@-13,f2=425@-19,c=300/1000-300/1000;
FXO
Account Active: Yes
Primary SIP Server: AsteriskIP
SIP User ID: 300
Authenticate ID: 300
Authenticate Password: YOUR PASSWORD
Name: PSTN
Dial Plan: { x+ | *x+ }
Local SIP Port: 5062 (Same as the Trunk, usually 5061)
Caller ID Scheme: Bellcore/Telcordia
Gain: TX & RX: +6db
FXO Termination
Enable Current Disconnect: No
Current Disconnect Threshold (ms): 200
Enable PSTN Disconnect Tone Detection: Yes
PSTN Disconnect Tone: f1=425@-19,f2=425@-19,c=150/150;
AC Termination Model: Country-based (GREECE) (Change it to your Country)
Channel Dialing
DTMF Digit Length (ms): 100
DTMF Dial Pause (ms): 100
First Digit Timeout (sec): 10
Inter-Digit Timeout (sec): 5
Wait for Dial-Tone: No
Stage Method (1/2): 1
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