Can I call from FXS1 port to FXS2 port?

Yes, they can communicate with each other by dialing the respective extension number. Both FXS ports need a valid sip account registered on the server. Route call automatically and transparently to PSTN line according to user configuration

Can I use a repeater with the DP715 to increase the coverage?

The DP715 cannot be used with a repeater to increase the coverage range.

Can we prevent random calls from any IP by limiting to accept calls from the SIP server only?

Yes. A user can enable a feature called "Allow Incoming SIP Messages from SIP Proxy Only". This field can be found under the FXS port configuration page in the web-gui.

Can you explain the use of 'PSTN pass through' and 'FXO port'?

PSTN Pass through port:
What it can do:
Local manual switching between PSTN and IP mode on a per call basis.
User can switch to PSTN line by pressing *00 (or the configured strings) for each call before they are placed. The device will revert back to the default IP mode once the phone is hung up.
It can allow a PSTN call to ring/call the phone connected to the FXS port.
It also serves as a life line in case of power outage.

What it CANNOT do:
Terminate a VoIP call into the PSTN port
Allow a call from PSTN to route other VoIP devices (different from the FXS phone) over the IP network
Automatically route calls made by the local user to PSTN line
Note: On the HT486 Rev 1.0, the PSTN port is only a life line port that switches to the PTSN network only when there is a loss of power.

FXO port:
It can support all the functions of a PSTN pass through plus:

Terminate a VoIP call into the PSTN port
Allow a PSTN call to call either the FXS phone or other VoIP devices over the IP network
Route call automatically and transparently to PSTN line according to user configuration

Does Grandstream allow HTTPS Provisionig with Grandstream Internal Certificate?

Yes. Grandstream allows HTTPS provisioning with Grandstream's very own certificate. If the server is configured to require a certificate for authentication, our devices can send an internal certificate that is flashed on all Grandstream devices. If you would like to use this feature please get in contact with our Grandstream Support for the GS Certificate.
Note: This is not the same as the "HTTP/HTTPS" user/password, which in this case this option is for authentication only once the connection has been established.

How can I disable hook/flash?

On the Advanced Settings page, there is a field On-Hook Threshold. One on the selections for this field is 'Hook/Flash OFF', this option will disable hook/flash on the phone connected to the ATA. To switch to a second channel, press FLASH button on the phone, instead of doing hook/flash. Note: This feature is not available on older hardware revision models.

How do I access the Web Configuration pages (for HT486/496/488)?

Please disconnect all connections to the HT486/HT496/HT488 and follow the instructions below:
:
Connect the analog touch-tone phone to the HT.:
Connect power supply.:
Connect the Ethernet cable between the INTERNET Source (ex. Router, Modem) and the WAN port on the HT.:
Connect another Ethernet cable between your PC and the LAN port on the HT:
Wait for 30 seconds till your PC gets an IP ADDRESS (192.168.2.2):
Now, open Internet Explorer and type in 192.168.2.1, you should see Grandstream Login Screen pop up.:
Enter 'admin' as the password:
Go to Advanced Settings page and switch "Enable WAN port HTTP access" to YES, hit 'Update' and then 'Reboot'.:
Disconnect your PC from the LAN port and connect it to any other port on your Router within the same LAN Segment:
10. Type in the actual IP ADDRESS of the HT (You can look this up by pressing *** on the phone, and then 02) on Internet browser, access the Web Configuration page as you did earlier and configure the device by filling in the information given by your Internet Telephony Service Provider (ITSP).

How do I configure my HT503 to call me when my Grandstream IP camera's alarm is triggered?

Click here to learn how!

How do I ensure my 911 calls are directed over the PSTN network and implement a dial plan for specific area codes without dialing a (1) prefix?

Create the following string under Dial Plan Configuration option under FXS PORT configuration page:
{L: 404x+| L:770x+ | L:678x+| L:911 | x+}
Please note that only HT503 supports this feature.

How do I make or receive PSTN calls on my HT (for HT486/386/488)?

When receiving a call, the phone connected to the HT simply rings. When placing a call, dial the 'PSTN Line Access Code' first, as configured on the Web Configuration Page (by default it is *00), and then dial the desired PSTN number.

How do I perform attend transfer?

Attend Transfer from A to B through HT:

A calls HT
HT talks to A
HT presses FLASH or does hook/flash to get new dialtone.
A is on Hold
HT calls B
HT talks to B
HT hangs up to perform the Attend Transfer.
A and B are in call now.

How do I setup a 3-way conference call?

Setting up a 3-way conference calling between parties using an HT, A and B is easy:

HT calls A
HT talks to A
HT presses FLASH or hook/flash and gets a new dialtone
A is on Hold
HT dials *23 and number for B
HT talks to B
HT presses FLASH or hook/flash to initiate the 3-way calling

How does the DTMF negotation work on the HT5xx series?

How the DTMF negotiation works: DTMF method negotiation: As a Caller/Callee: 1. If disable DTMF negotiation, DUT will use the first dtmf method from webUI. 2. If enable DTMF negotiation, DUT will check if there is RFC2833 from the SDP of 200OK (for Caller) or INVITE (for Callee). If it has RFC2833, DUT will check if itself has RFC2833 in DTMF list. If DUT also has RFC2833, then RFC2833 is chosen. Otherwise, DUT will check if there is SIP INFO supported by callee/caller. And then check itself. If DUT also support SIP INFOin DTMF list, then DUT uses SIP INFO. Otherwise, DUT directly use IN_AUDIO.

Note: If negotiation is enabled, the priority of DTMF order doesn’t matter. DUT will first try to match RFC2833 and then SIP INFO, the least priority is In-Audio. If negotiation is disabled, DUT will use the first DTMF method configured in webUI.

What are the main differences between the 9 models of Handytones?

What format does the Grandstream Config Generator encrypt the file into?

Our configuration files are encypted with 128 AES. As for the new XML format, it will be decrypted with 256 AES.

What hardware revision is my Handytone?

At the bottom of the device, there will be a white sticker. On this sticker will be a note:
'Rev: x.0' where x is the Hardware Revision:
You can also get this information under the Status tab of the Web Configuration pages.:
Your hardware revision is very important information. Depending on your hardware revision, it may/may not be able to upgrade to the latest release.:
Ex. HT286 Rev 2.0 upgrades up to 1.0.7.19 firmware, while Rev 3.0 can upgrade to the latest release.:
Note: Once there is a new hardware revision out in the market, the older revision is no more manufactured or sold.

What is a dial-plan and how do I configure it?

The dial-plan is a set of rules that governs the call-routing behavior of a device. When a user dials a sequence of number the device will refer to the rules in the dial-plan in order to determine how best to connect the call. Users can refer to the user manual for more details on how to configure the dial-plan. All user manual for your products can be found here

What is the led pattern of a HT502/503?

- Power led ON indicates power is connected
- WAN led ON indicates port activity
- LAN led ON indicates PC (or LAN) port activity
- Phone1/2 or Line led indicates status of the respective FXS port or FXO in case of HT503. ON is Busy, OFF is available and slow blinking there is a voice mail for that port.
Slow blinking of WAN and LAN together means the product’s firmware is in upgrading or provisioning state.

Which phone will ring if there is an incoming call on my wire (fixed) line (HT386)?

The Phone connected to FXS1 port will ring when there is an incoming call on the Fixed Line.