Grandstream HT-503

Content

About

Version 1.0.14.1

Used Symbols

About

The HT503 is a hybrid Analog Telephone Adapter (ATA) and VoIP Router. It features both an FXS (analog telephone) port and an FXO (PSTN) port so you can have backup lifeline support using a PSTN line in the case of a power outage. It also integrates 2 10/100 RJ45 ports with an integrated high performance NAT router.

Gateway web page

link
Default loginN/A
Default passwordadmin

Version 1.0.14.1

Version detail : Program -- 1.0.14.1, Bootloader -- 1.0.0.18, Core -- 1.0.14.1, Base -- 1.0.14.1, Extra -- 1.0.14.1, CPE -- 1.0.1.48

Note

For correct function Enhanced Integration license for 2N IP Intercom is needed.

Tested Functions

 Voip-to-PSTN calls

2N IP Intercom

2.19.0.28.5

(tikot)

Voip-to-PSTN calls Settings

In Directory / Users  create user for HT503 and in field Phone Number write down IP address of HT503 with port 5060.

In next step assign this created user to number of buttons corresponding to the number of phone numbers that will be called from 2N IP Intercom. In Hardware / Buttons.
Afterwards it is needed to create Function in Automation for dialing up numbers in Analog PBX. Function is as follows:

  1. Event for pressing the quickdial button
  2. Delay for 1 s
  3. Dialing phone number with DTMF, which is ended # (this symbol is used to confirm the number).

In following example the first button will dial number 400, second will dial 401 and the third dials 402. The following function is avaible for download here.

Finally the DTMF should be set to sending and accepting DTMF RFC-2833.

Connection for this scenario should be following:

The 2N IP Intercom should be connected to the WAN port of HT-503 (via Ethernet Switch, it doens't need to be connected to internet or on the public IP address), the HT-503 should be connected with PBX with Line port (FXO) to FXS port on PBX.

On the side of Grandstream HT-503 following settings should be set in Basic settings,(sip server is IP adress of 2N IP Intercom, and user ID can be any number, port should be 5060):

In advanced settings following should be set, other settings should remain from factory settings:
In section FXS PORT deactivate SIP account:
And finally following should be set to FXO PORT section of settings (SIP server and proxy is 2N IP Intercom, and SIP User ID is any number):
In the end it is important to set correct Hang-Up tone (PSTN Disconnect Tone) for correct termination of Call, this should be set according to your PBX:

Used Symbols

(tikot) - Compatible

(varování) - Work with limitation

(chyba) - Incompatible