Unable to make outgoing calls on HT488

[newlifecc]

I am setting up my first 3CX. Everything is working except - I cannot seem to find the correct configuration to make outgoing calls through my Grandstream HT488. I have my calls setup with a prefix of "0", strip "1" digit and prepend "*00".

- I am not getting an error through the voip client.
- With a spy phone, I only hear the dial tone, no connection or attempt to dial out.
- I can dial out with a phone connected to the HT488, when I dial *00.

I have seen several posts about this. Has anyone found a solution?

Thanks so much!

 

[Halea]

I'm certainly not the best qualified person on this forum to give you advice as I am struggling myself with HT488's FXO circuit :|
Incoming PSTN works, outgoing PSTN doesn't
That said, I don't think that you need to prefix the number that you're calling with *00. I believe that *00 is only to direct connect FXS to FXO and get a dial tone to the locally connected phone. And I hope that I'm not doing a major blunder here :lol: I actually never tried to prefix my calls with *00. It just didn't occur to me. Where did you get the idea that it needs to be appended to the call number? Just curious.
Halea

 

[newlifecc]

I'm not sure where I got it, other than just a recognition that it needs something. I think the answer is found in these two resources:

Options at bottom of Basic Settings page (Graphics won't show here)

PSTN access code: (key pattern to use PSTN line, default is "*00")
PIN for PSTN Calls: (Enter digits to authorize calling PSTN numbers from VOIP, no default)
PIN for VOIP Calls: (Enter digits to authorize calling VOIP terminals from PSTN, no default)
Route Call to PSTN: Outbound calls will be routed to PSTN port when dialed digits match one of the following:
Forward to PSTN: (VoIP calls will be forwarded to the specified PSTN number if ring no answer)
Forward to VoIP: (PSTN calls will be forwarded to the specified VoIP number)
FXO One Stage Dialing: No Yes (No-default, SIP UserID is used as PSTN number if set to Yes)

HT-488 Users Manual

5.2.9 VoIP-to-PSTN Calls
This function is applicable on FXO port that functions as a bridge between VoIP and PSTN. The user
can remotely use PSTN line to initiate a call.
To make a VoIP-to-PSTN call:
1. Dial the FXO SIP account phone number to establish the VoIP session. The caller will hear the
ring back tone once. Then the caller hears either a special continuous tone or a dial tone. The
special continuous tone is played if the pin code is configured, or the dial tone otherwise.
2. Enter in the pin code that is configurable on the configuration page. The caller will hear the dial
tone and get connected to the PSTN line if the pin code is valid, otherwise the continuous tone
is played again to prompt caller to enter in the pin code again. The use may try up to 3 times to
enter in pin code, if none is valid, HT488 will hang up.
3. After the caller hears dial tone from PSTN line, the caller can start dialing number to make
calls.
Note:
• Users can choose whether apply password protection for VoIP-to-PSTN calls or not. A PIN
(Pin for PSTN calls) consists of up to 8 numeric digits can be configured through BASIC
SETTINGS of the web configuration page. By default, there is no password protection, i.e.
there is no authentication required for callers on the use of PSTN line through HT488.
• When a PIN is configured for VOIP-to-PSTN call flow, the VoIP device that calls into the
HT488 FXO account needs to configure RFC2833 or SIP Info for DTMF digit transmission.
• Upon hearing the special continuous tone for PIN code input, if the caller don’t enter any digit,
HT488 will time out and hang up the call in 10 seconds. During any stage of DTMF digits
input, a 4 seconds timeout is applied to serve as an end of PIN or destination number input.
Users may also use the “#” key to indicate the end of an input.
• On the web configuration page, if the “Forward to PSTN” is configured, the second stage
dialing is eliminated, i.e., after dialing into the FXO SIP account number, the PSTN number
will be called automatically.

There is some combination needed. The line that catches me in the manual is

1. Dial the FXO SIP account phone number to establish the VoIP session.

There is soemthing 3CX needs to do to access the FXO port.

Thanks for your reply!

 

[aguilar]

hello
did you tried just prepend 0? wich is the same as remove 0 and add 00

regards

 

[newlifecc]

Thanks for the suggestion. No go so far. It does not seem that 3CX is communicating at all with outgoing calls to the unit; although everything appears to be working on the 3CX side - it dials w/o error. When I listen to the PSTN line, I hear no attempt to connect.

Thanks again!

 

[Halea]

I've just figured out the reason for which my HT488 didn't call out to PSTN :idea: , see if it works for you. On the FXO tab, all the way down the list check if you have "Enable Current Disconnect" set to "Yes"; if that's the case then increase the value for "If current disconnect enabled, use threshold" from 100 mS to 3000mS.
I'm now positive that you don't need to dial *00 or anything else. Stick to the 3CX recipe for all the other settings.
Good luck.
Halea

 

[newlifecc]

Still no success here. What version of firmware are you on? I am on

Program-- 1.0.3.96 Bootloader-- 1.1.0.1 HTML-- 1.0.3.96 VOC-- 1.0.0.13

I still does not sound as if it is picking up the line. Grrrr
Congradulations on your success! and thanks for your update!

 

[Halea]

By the way, is the phone from which you're trying to call out on PSTN attached to the FXS port of HT488? If that's the case, it might be a whole different issue. First, try another extension or port and ensure that you get to call out over HT488's PSTN. The HT488 is a box full of (unpleasant) surprises.
I have my own saga with HT488, if you read it you'll see what I went through so far. It's not so bad, but I am fuming on the countless hours that I spent troubleshooting the crap :x
:roll: Halea