Grandstream HandyTone 502 - HT502
From VoIP.ms Wiki
The Grandstream HandyTone 502 is a two-line analogue telephone adapter with a built-in single port wired 10/100Mbps LAN router and network address translation (NAT). It is reported to support both tone and pulse dial (a capability which was removed from the later HT702).
A product description and manual is available from the manufacturer. You can access your User Manual here and it includes your device's star codes on page 25: http://www.grandstream.com/products/ht_series/ht502/documents/ht502_usermanual_english.pdf
Connecting the HandyTone
1. Connect a standard touch-tone analog telephone to PHONE port (or PHONE1, PHONE2 port for HT386/496/502).
- Connect a PSTN telephone line to LINE port (optional, applies to HT386/486/488/503 only).
3. Insert the Ethernet cable into the Ethernet port (HT286/386) or WAN port (HT486/488/496/502/503) of HandyTone and connect the other end of the Ethernet cable to an uplink port (a router or a modem, etc.).
4. Connect a PC to the LAN port of HandyTone (optional, applies to HT486/488/496/502/503 only).
5. Insert the power adapter into the HandyTone and connect it to an electrical outlet.
6. Using the HandyTone embedded web server or IVR (Interactive Voice Prompt) menu, you can further configure the phone using either a static IP or DHCP.
Accessing to the Web Configuration
1. From the analog phone, press *** to get into the IVR menu. Enter option 02 to obtain the HandyTone’s IP address.
2. For HT486/488/496/502/503, please enable the “WAN side HTTP access” option by entering IVR option 12 and press 9. A reboot or power cycle of the HandyTone is required after this change. You can also access the HandyTone’s web configuration from a PC connected to the LAN port via 192.168.2.1.
3. Type the HandyTone’s IP address in your PC browser.
4. Log in using password “admin” to configure the HandyTone.
Click on FXS 1 to configure your first line.
Fill the followings fields.
- Account Active: Yes
- Primary SIP Server: atlanta.voip.ms (Pick one of VoIP.ms multiple VoIP Servers)
- Outbound Proxy: Set the same server you've configured at the Primary SIP Server field
- NAT Traversal: No, But Keep-alive
- SIP User ID: 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)
- Authenticate ID: 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)
- Authenticate Password: ********* (account password)
- User ID is phone number: No
- SIP Registration: Yes
- Register Expiration: 5 minutes
- Preferred Vocoder: Select as primary option the PCMU codec and as secondary option the G729 codec
Don't forget to reboot your device after you've applied the recommended settings.
Known Issues and Resolutions
Hearing an Echo on the Line:
Please go to your FXS port setting screen, the one you are using with our service, and verify that the option 'Disable Line Echo Canceller' is set to NO. You can also adjust the setting Gain, where the Rx (Other Person) is a gain level for signals transmitted by FXS and Tx (You) is a gain level for signals received by FXS.
Receiving Weird Calls such as from CallerID 100 or in the middle of the Night not showing in your CDR:
These calls are not going through our Network but rather through the internet directly to your ATA Device. Please check in the Web GUI under each FXS port tab that this option is enabled: Allow Incoming SIP Messages from SIP Proxy Only: Set to YES. This will make sure you can only receive calls from VoIP.ms.