HELLO & WELCOME

[WelshPaul]

Below is my Grandstream HT812 (Firmware v1.0.33.4) configuration for use with Andrews & Arnold (AAISP) here in the UK.
(Replace +1234567890, 123456 and 01792 with your own unique parameters.)

BASIC SETTINGS

• Time Zone: GMT (London, Great Britain)
• Self-Defined Time Zone: GMT0BST,M3.5.0/1,M10.5.0

ADVANCED SETTINGS

• STUN server is: stun.aa.net.uk:3478
• Keep-alive Interval: 20
• Firmware Upgrade and Provisioning: HTTP
• Firmware Server Path: firmware.grandstream.com
• Automatic Upgrade: Yes, every 10080 minutes(30-5256000).
• Always Check for New Firmware at Boot up: Yes
• System Ring Cadence: c=400/200-400/2000;
• Dial Tone: f1=350@-19,f2=440@-22,c=0/0;
• Ringback Tone: f1=400@-20,f2=450@-20,c=400/200-400/2000;
• Busy Tone: f1=400@-20,c=375/375;
• Reorder Tone: f1=400@-20,c=400/350-225/525-0/0;
• Confirmation Tone: f1=1400@-10,c=0/0;
• Call Waiting Tone: f1=400@-20,c=100/2000;
• Prompt Tone: f1=350@-19,f2=440@-22,c=0/0;
• Conference Party Hangup Tone: f1=400@-20,c=0/0;
• Special Proceed Indication Tone: f1=350@-19, f2=440@-22, c=750/750-0/0;
• NTP Server: time.aa.net.uk

PROFILE 1

• Primary SIP Server: voiceless.aa.net.uk
• Outbound Proxy: voiceless.aa.net.uk
• NAT Traversal: Keep-Alive
• Unregister On Reboot: Yes
• Outgoing Call without Registration: No
• Register Expiration: 1
• OPTIONS/NOTIFY Keep Alive: OPTIONS
• Check SIP User ID for incoming INVITE: Yes
• Allow Incoming SIP Messages: Yes
• MWI Tone: Special Proceed Indication Tone
• Dial Plan: { 10[015] | 11[129] | 15[01] | 999 | 1571 | 17070 | 11[68]xxx | 08001111 | 0845464x | <=01792>[2-9]xxxxx | <141=14101792>141[2-9]xxxxx | <1470=147001792>1470[2-9]xxxxx | 0[1235789]xxxxxxxxx | 1410[1235789]xxxxxxxxx | 14700[1235789]xxxxxxxxx | 00xxx. | x+ | \+x+ | *x+ | *xx*x+ }
• Preferred Vocoder (in listed order):
  • choice 1: PCMA
  • choice 2: PCMA
  • choice 3: PCMA
  • choice 4: PCMA
  • choice 5: PCMA
  • choice 6: PCMA
  • choice 7: PCMA
  • choice 8: PCMA
• SLIC Setting: UK
• Caller ID Scheme: SIN 227 - BT
• Hook Flash Timing: In 40-2000 milliseconds range, minimum: 60 maximum: 200
• Ring Frequency: 25Hz
• Ring Tone 1: c=400/200-400/2000;
• Ring Tone 2: c=400/200-400/2000;
• Ring Tone 3: c=400/200-400/2000;
• Ring Tone 4: c=400/200-400/2000;
• Ring Tone 5: c=400/200-400/2000;
• Ring Tone 6: c=400/200-400/2000;
• Ring Tone 7: c=400/200-400/2000;
• Ring Tone 8: c=400/200-400/2000;
• Ring Tone 9: c=400/200-400/2000;
• Ring Tone 10: c=400/200-400/2000;
• Call Waiting Tone 1: f1=400@-20,c=100/2000;
• Call Waiting Tone 2: f1=400@-20,c=100/2000;
• Call Waiting Tone 3: f1=400@-20,c=100/2000;
• Call Waiting Tone 4: f1=400@-20,c=100/2000;
• Call Waiting Tone 5: f1=400@-20,c=100/2000;
• Call Waiting Tone 6: f1=400@-20,c=100/2000;
• Call Waiting Tone 7: f1=400@-20,c=100/2000;
• Call Waiting Tone 8: f1=400@-20,c=100/2000;
• Call Waiting Tone 9: f1=400@-20,c=100/2000;
• Call Waiting Tone 10: f1=400@-20,c=100/2000;

FXS PORTS

• SIP User ID: +1234567890
• Authenticate ID: +1234567890
• Password: 123456
• Enable Port: Yes

Using Multiple Phones Behind the Same Router?
If there's more than one VoIP phone/device on the same local network, assigning each device with a different local UDP port will avoid port conflicts.

As I use multiple devices, I have changed the following parameters to suite my needs:

PROFILE 1

• Local SIP Port: 46260
• Local RTP Port: 46204

If you don't use multiple VoIP phone/devices on the same local network, leave the above at their default settings as shown below:

PROFILE 1

• Local SIP Port: 5060
• Local RTP Port: 5004

Screenshots of my configuration below:

[tonygibbs16]

Thanks very much for this setup guide @WelshPaul

I have the same ATA and firmware as you, but I am using Sipgate Basic.

This guide really helped me get setup and make my ATA sound UK-like.

Blessings,
Tony