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VictorienXP / Grandstream-HT813

Setup of Grandstream HT813 with PSTN line in France and FreeBPX

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Grandstream HT813 - France PSTN + FreePBX

This is notes of my settings for using a Grandsteam HT813 with a french PSTN line with FreePBX.

Manual https://www.grandstream.com/hubfs/Product_Documentation/HT813_Administration_Guide.pdf

FreePBX: 15.0.17.24

Grandstream HT813: 1.0.9.1

Overall setup

FX0 port <--> PSTN line (tested with France Telecom POTS and Freebox Revolution router phone line)

FXS port <--> analog phone

WAN port <--> your network

LAN port <--> whatever device... it will be bridged

I assume that your FreePBX use PJSIP on port 5060 UDP.

In the exemple the phone number is 0123456789.

Setup in FreePBX

Analog Phone Extension

Create an extension PJSIP for the analog phone. For the exemple I will use the extension 3210 with secret your_strong_password_with_letter_and_number_for_extension_3210

Trunk Setup

I use a SIP chan_pjsip Trunk.

General

Trunk Name: 0123456789
Outbound CallerID: <0123456789>
Asterisk Trunk Dial Options (Overide): TR

The overide on Asterisk is in order to have a tone when outbound call are placed... there is a 10sec silence delay until the PSTN is actually connected (any idea on how to reduce that would be great).

pjsip Settings - General

Username: 0123456789
Secret: your_strong_password_with_letter_and_number_trunk
Authentication: Outbound
Registration: Receive
Context: from-trunk

pjsip Settings - Advanced

DTMF Mode: RFC 4733
Trust RPID/PAI: Yes

pjsip Settings - Codecs

alaw

I disabled all the other codecs... somehow I ran into troubles when call were coming in.

HT813 Setup

Find the device on your network and access it via http. Note: when I enabled https, I got into certificat trouble and I had to reset my HT813.

Default access: admin/admin

Basic Settings

  • Change End User Password

  • Change Viewer Password

    Disable SSH: Yes Time Zone: GMT+01:00 Paris Device Mode: Bridge Reply to ICMP on WAN port: Yes Enable LAN DHCP: No Unconditional Call Forward to VOIP User ID: 0123456789 Sip Server: IP_of_FreePBX Sip Destination Port: 5060

Advanced Settings

  • Change Admin Password

If you have a VLAN for your VOIP network then you can set it here. Be sure to have a DHCP server there or setup a fixed IP address.

802.1Q/VLAN Tag: your_VLAN_ID

Under Firmware Upgrade and Provisioning, empty the Config Server Path. I advise you also to enable Automatic Upgrade.

3CX Auto Provision: No (for added security)
Enable TR-069: No (for added security)
System Ring Cadence: c=1500/3500;
Dial Tone: f1=440@-10,f2=0@-10,c=0/0;
Ringback Tone: f1=440@-10,f2=0@-10,c=1500/3500;
Busy Tone: f1=440@-10,f2=0@-10,c=500/500;
Reorder Tone: f1=440@-10,f2=0@-10,c=50/50;
Confirmation Tone: f1=350@-11,f2=440@-11,c=100/100-100/100-100/100;
Call Waiting Tone: f1=440@-13,c=300/10000;
Prompt Tone: f1=350@-13,f2=440@-13,c=0/0;

Lock Keypad Update: No
Disable Voice Prompt: Yes
Disable Direct IP Call: Yes
Life Line Mode: Auto
NTP Server: 2.fr.pool.ntp.org

If you want to debug your HT813, then you might want to setup syslog. You can use this software to setup a syslog server for Windows with a nice GUI. https://github.com/MaxBelkov/visualsyslog/tree/v1.6.4

FXS Port

Primary SIP Server: IP_of_FreePBX
Outbound Proxy: IP_of_FreePBX
SIP User ID: 3210
Authenticate ID: 3210
Authenticate Password: your_strong_password_with_letter_and_number_for_extension_3210
Register Expiration: 5
Allow Incoming SIP Messages from SIP Proxy Only: Yes
SIP REGISTER Contact Header Uses: WAN Address
Use First Matching Vocoder in 200OK SDP: Yes
SLIC Setting: EUROPEAN CTR21
Gain: RX 0dB

FXO Port

Primary SIP Server: IP_of_FreePBX
Outbound Proxy: IP_of_FreePBX
SIP User ID: 0123456789
Authenticate ID: 0123456789
Authenticate Password: your_strong_password_with_letter_and_number_trunk
Register Expiration: 5
Allow Incoming SIP Messages from SIP Proxy Only: Yes
SIP REGISTER Contact Header Uses: WAN Address
Preferred Vocoder choice 1->7: PCMA
Voice Frames per TX: 10
Caller ID Transport Type: Relay via SIP P-Asserted-Identity
Gain: TX +2dB RX 0db
Enable PSTN Disconnect Tone Detection: Yes
PSTN Disconnect Tone: f1=440@-30,f2=440@-30,c=500/500;
AC Termination Model: Auto-Detected
Number of Rings: 2 (if you put 1, the system will not get Caller ID)
PSTN Ring Thru FXS: No
Stage Method (1/2): 1

Once all those setup done, you can reboot your HT813. The FX0 and FXS led should light up once the lines are registered in FreeBPX. You can also check the status in the Status page.

Anonymous Caller

It seems that with my settings the option Anonymous Call Rejection is not working... somehow when an anonymous arrive, it takes the line number as Caller ID (exemple: 012345789).

The trick I use is to setup a new Inbound Routes in FreePBX with the following settings:

DID Number: 0123456789
CallerID Number: 0123456789

Gain and echo problems

If you have gain or echo problems, you should modify your gain settings in the FXS/FXO pages. The change of gain listed above are for my phone lines and might not apply to you.

Credits

Here are the ressources which helped me to compile those settings.

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Setup of Grandstream HT813 with PSTN line in France and FreeBPX