Raspberry Pi 3CX with Grandstream Handytone HT 503

[CountMagnus]

Do not know if anyone can help.

I am setting up a Raspberry Pi with 3CX. I have set up trunks to Sipgate and added two extensions VVX411 and VVX300.All works fine.

Therefore the next step was to connect the PSTN line via the HT503,

The HT503 has firmware is Program -- 1.0.16.2 Bootloader -- 1.0.0.18 Core -- 1.0.16.1 Base -- 1.0.16.2 Extra -- 1.0.16.1 CPE -- 0.0.1.97

and I followed the following forum post.

However I just can not get it to work.

I have a "new" extension showing that I was not expecting

The SIP trunk is not registered

and the port status on the HT503 of the FXO is busy and not registered.

Could anyone provide any guidance or help. I am a bit stuck and running out of ideas.

Would anyone have some screenshots of how they would set this up.

POST - With info that I followed
Please follow this steps for HT503:
- Upgrade the HT503 to f/w 1.0.4.100
- Reset the device to factory default
- Make sure you use a fixed IP-address for the HT503
- HT503 and 3CX PBX must be on the same LAN
- Setup in “Basic Settings” the right Time Zone
- Setup in “Advanced Settings” the right Call Progress Tones
- De-activate FXS port
- Setup in “FXO Port” your country settings and…
FXO Termination
- Number of Rings: 2
- PSTN Ring Thru FXS: No
- PSTN Ring Thru Delay (sec): 1
Channel Dialing
- Wait for Dial-Tone: No
- Stage Method (1/2): 1

Now in 3CX please follow next steps:
- Create an extension/IVR
- Create Generic PSTN Gateway
- Gateway IP-adres: <HT503>
- Gateway Port: 5062
- During creation, setup Inbound Route to previous created ext/IVR
- Copy SIP ID, Auth ID and Password to HT503 FXO (type in, do not copy/paste!!!!)

Now setup “Basic Settings” HT503
- Unconditional Call Forward to VOIP: <previous created ext/IVR>@<Local IP PBX>:5060

Check if HT503 is registered

 

[YiannisH_3CX]

Gateway Port: 5062

Are you sure the FXO port is set to 5062? The default should be 5060 so check that. Since we do not support the device we cannot really suggest what the correct configuration should be to get it to register

 

[leejor]

With most PSTN Gateways, you would create a generic trunk and register the FXO port to that. Reading some of the other posts dealing with the HT503, it would seem that others have found that does not work. Are you attempting to register as a trunk, or as an extension? What does the 3CX Activity log show, is there a registration attempt? You say the trunk is not registered.

Going by your post, i assume you came across...
https://www.3cx.com/community/threads/grandstream-handytone-ht503-fxo-gateway.44492/

and...

https://www.3cx.com/community/threads/grandstream-ht503-fxo.40718/

I have not used the 503, but I am familiar with the SPA3102 gateway

 

[CountMagnus]

Hi,

Many Thanks for any help with this.

To clarify how the network is set up. HT503 and 3CX PBX are on the same LAN. They both have static address outside the DHCP range. The main router is a normal Home router connected to the internet via a static IP.

The instruction to "Setup in “Advanced Settings” the right Call Progress Tones" I am not clear on so I have not changed any of these settings.

When setting up the 3CX it said to:-

1/ Create an extension/IVR - I have existing extension set up and working to a VVX411 and a VVX300 so have not changed anything. Do I need to set up another Extension and if so how do I sent this up as the software appears to want to set up the extension to type of phone from its database.
2/ Create Generic PSTN Gateway - I set this up as the generic but it says to Copy SIP ID, Auth ID and Password. In the settings on the 3CX I only have Copy SIP ID and Password. Therefore I have assumed that the Copy SIP ID, and Auth ID are the same (10000) is this correct ?

On the HT 503 I have set the FXO port to port 5062 (Local SIP Port) and in the 3CX Generic PSTN Gateway I have it set to 5062..

I hope from the above info it may help you see my error and you can then provide some advise of were I am going wrong. Thanks

 

[complex1]

Hi,

Before you start with the configuration, upgrade the firmware of the HT503 to the latest one. (v1.0.16.3)
Then reset the HT503 to default settings. Not just a reboot, but reset it.

Create a new extension as you did for the VVX phones.
Then under SIP Trunks, Add gateway and select Generic and fill in what's needed.
Now copy/paste all relevant data to the FXS and FXO tabs in the HT503.
Do not forget the setting as I post before. (Jun 4, 2016... Wouw)

 

[CountMagnus]

Hi,

Many thanks for the help.

Have upgraded to v1.0.16.3

The bit I can not figure out is how to set up a "new" extension. When I set up a new extension i need to provide the Mac address and the model. It says add phone in a pop out type windows and say "Choose from available models" and the provide Mac address. I am not clear what Model to choose from. I have tried extension 200. (All mine are in the 100-199 range) and a Grandstand model.

Is this what I need to do ?

If I reboot the HT503 a new extension does show with the correct Mac address but the extension number is "new"

Any ideas ?

 

[complex1]

Create an extension as follows:
Log in the Management menu, click Extensions, click Add.
Set the desired Ext nr., First Name & Last Name, click OK, extension is created.
Open the new extension, go to Phone Provisioning tab and copy/paste Auth ID and Password in to the HT502 FXS tab.
The HT503 will register to the PBX.
Almost the same you can do for the FXO part of the HT03, but then you have to create a gateway first.

 

[leejor]

Therefore I have assumed that the Copy SIP ID, and Auth ID are the same (10000) is this correct ?

The SIP ID will probably be the trunk number, but, I'm not certain that the Auth Id will be the same.The 3CX trunk settings should confirm the Auth ID.

Does the 3CX Activity Log show the HT503 attempting to register the trunk with 3CX?

 

[CountMagnus]

Hi,

I have added some further configuration and have now goy the HT503 to show that both the FXO and FXS ports are both registered.

However I still can not get any calls through.

I also plugged a phone into the port and checked for any side tone. It was dead.

In the HT 503 manual it says.

"If the 503 loses power or lost registration with SIP server, device will switch to mode when PSTN
line will be transparently connected directly to phone connected to FXS port. It will function as a
jack, enabling a direct connection to the PSTN Line."

I can not get this to work. Therefore does this suggest that there could be a problem with the unit ?

 

[complex1]

In which port did you plugged in the (old fashion analog) phone?
FXS or FXO?
If you use the FXS port then you can test is using the echo test (*777)

 

[complex1]

If you want to use the HT503 as FXO gateway make next settings:

FXO Termination
- Number of Rings: 1
- PSTN Ring Thru FXS: No
- PSTN Ring Thru Delay (sec): 2
Channel Dialing
- Wait for Dial-Tone: No
- Stage Method (1/2): 1

Basic Settings
- Unconditional Call Forward to VOIP: <SIP User ID>@<IP addr of PBX>:5060

Create outbound route to PSTN Gateway

 

[CountMagnus]

Many Thanks for your help but I can not get any Side Tone from the FXS port when a phone is plugged in. It should provide something when the power fails from the PSTN line. I think the unit is faulty. I am going to test further and reset it to see if I can get it to work with the basic functions that it has without the 3CX and use just a connected phone on the FXS port and a PSTN line on the FXO port.

 

[leejor]

If the 503 is not powered, then, yes, a phone plugged in the FXS port should get dial tone from a phone line connected to the FXO port. If you don't get that, it may be a faulty power fail relay (you should hear it click when you power up, and remove power), or, it might be a faulty phone cord.

 

[fiachra]

If you want to use the HT503 as FXO gateway make next settings:

FXO Termination
- Number of Rings: 1
- PSTN Ring Thru FXS: No
- PSTN Ring Thru Delay (sec): 2
Channel Dialing
- Wait for Dial-Tone: No
- Stage Method (1/2): 1

Basic Settings
- Unconditional Call Forward to VOIP: <SIP User ID>@<IP addr of PBX>:5060

Create outbound route to PSTN Gateway

I am having a problem with caller ID on the analog phone on the FXS port. I have followed the above advice, but caller ID just shows 10000 (3CX extension number). If I change Number of rings to 2 then the caller ID appears correctly….BUT the FXO line rings 5-6 times before the call appears on the FXS line and the ringing starts. By this time some callers have given up and already hung up!.
My HT593 firmware is 1.0.16.3.
So the FXS phone either rings immediately with no caller ID, or waits for 5-6 rings before sending the call to the FXS.
Has anybody else had this issue? Any suggestions welcome.
Thank you

 

[leejor]

Because Caller ID (if using Bellcore type), is sent between the first and second ring, then the call cannot be "passed on" to 3CX before the data is collected, or you will simply see the trunk number, as you've experienced.

The option PSTN Ring Thru Delay, set to 2, or 3 seconds, should accomplish this. If there is another delay, caused by how the call routes in 3CX, then, yes, a caller may hear 5, or more rings before someone answers.

Check the 3CX Activity Log, after a test call, to see how long it takes for your extension to ring after 3CX gets an INVITE from the gateway.

Looking again at the options, that ring thru delay may be for the FXS port on the device, refering to the option above.. There may be some other option to delay sending the call to 3CX.

 

[leejor]

Looking again at this thread...

https://www.3cx.com/community/threads/grandstream-handytone-ht503-fxo-gateway.44492/

There may not be an option available, and that is why there is such a complicated way of getting this to work, as a trunk on 3CX.

 

[AWS2P]

I'm most certainly too late for this, but I found an old check list I had for myself, for steps to follow with HT503 and 3CX, I paste here if it may help :

Please follow this steps for HT503:
- Upgrade the HT503 to f/w 1.0.4.100 (this was fw in time of writting, so newer may be here now)
- Reset the device to factory default
- Make sure you use a fixed IP-address for the HT503
- HT503 and 3CX PBX must be on the same LAN
- Setup in “Basic Settings” the right Time Zone
- Setup in “Advanced Settings” the right Call Progress Tones
- De-activate FXS port
- Setup in “FXO Port” your country settings and…
FXO Termination
- Number of Rings: 2
- PSTN Ring Thru FXS: No
- PSTN Ring Thru Delay (sec): 1
Channel Dialing
- Wait for Dial-Tone: No
- Stage Method (1/2): 1

Now in 3CX please follow next steps:
- Create an extension/IVR
- Create Generic PSTN Gateway
- Gateway IP-adres: <HT503>
- Gateway Port: 5062
- During creation, setup Inbound Route to previous created ext/IVR
- Copy SIP ID, Auth ID and Password to HT503 FXO (type in, do not copy/paste!!!!)

Now setup “Basic Settings” HT503
- Unconditional Call Forward to VOIP: <previous created ext/IVR>@<Local IP PBX>:5060

Check if HT503 is registered

Now you can make and receive calls through PSTN.

Good luck.