- Fri 28th Jan 2022, 22:15
#6435
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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
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
PROFILE 1
(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
- 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
- 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;
- 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
PROFILE 1
- Local SIP Port: 5060
- Local RTP Port: 5004
Attachments:
VoipIT, tonygibbs16 liked this
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