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