HT503 can not dial out - debug syslog showing minimal info :(

bug

#1

We have installed many HT503 devices all over the country, all working fine (with ISDN or “analog” lines coming out from internet-routers provided by IPSs) with the SAME SETUP.
The device is only used for one purpose: recognize CallerID.
(Every other functions, like VOIP is turned off.)

But we’ve just tried the same system in an old restaurant, still using an old ANALOG line.
(And DSL internet over it, max 10/1 Mbit . The provider center is 1600 meter away, week signal, etc.)

Symptom:
We can receive incoming calls, but
Can not dial out through analog.

SYSLOG:

13:24:36.364 … Call(3)::dialPlanEngine, DialPlan: 0x4 0630480xxxx line: 0
13:24:40.373 … DialPlan timeout: 0x4 0630480xxxx
13:24:40.373 … DialPlan timeout: repetitive rule
13:24:40.389 … Call(3)::run, Dialing 0630480xxxx
13:24:40.389 … ATACtrl::call, cannot make the call, statusCode = 0, chan status =CALL_DIALED, emergency call =0
13:24:40.389 … EventManager::unregisterEventListener: listener Call
13:24:40.389 … Dispatching event: 17 (CALL_FAILED) on port 0:0
13:24:40.389 … SigCtrl::processCallFailed() on port 0:0 ptype 0 sCode 0, status CALL_DIALED , sigReferred 0
13:24:40.389 … ATACtrl::processCallFailed on port 0:0, status = CALL_DIALED/CALL_IDLE stCode:0 canConf:1 ,sigReferred:0
13:24:40.405 … Vinetic::stopTone, Stop tone on port 0, direction 0
13:24:40.405 … Vinetic::playTone, Play tone 36 on port 0, direction 1
13:24:40.405 … Call(3)::~Call, Deleting Call object 3 at port 0:0
13:24:43.696 … Dispatching event: 3 (PHONE_ON_HOOK) on port 0:0

As you can see the device is providing extra small amount of debug info about the actual call procedure. (Waiting for dial tone ... tone received ... generating DTMF "0" ... waiting 127ms ... generating DTMF "6" ...)

Of course if we plug the phone directly into the line (leaving HT503 out) it works perfect.

So HOW do I figure out, why is this device not dialing out in this specific situation?
I’m open for every idea.

PS: this new forum is much worth that the old one. :frowning: can not format [code]


#2

As it turns out, it was NOT an IPS problem.
The HT-503 could not dial out !
Currently I have an other device and managed to reproduce the same problem by:

  • Setting up everything and downgrading to 14.0.1.100 at the SAME TIME.
    (normally this is a separate procedure, first downgrading, than setting up.)

After this mistake (to try to save some time and do everything in one step) :

  • it is impossible to reset the device, it always keeps ALL settings.

Need a “clean” config file to try to overwrite the current one.


#3

I too use these quite a bit. I have no issue resetting them.

You did not state how the device is connected and if only used for inbound CID, what is the significance of being able to dial-out and how does one dial out, i.e., what is it connected to that is used to dial out? You already stated that using the phone you could dial out.

You will note that the FXO port is set to 5062 by default which implies that the device handling the call needs to send the INVITE to 5062.

The device uses an electronic off-hook mechanism which is a little tricky to use if the loop current is low. You may need to experiment with the disconnect threshold by setting it to 400 or so.


#4

Thank you for your help :slight_smile:

  • no, I can NOT dial out with the last 2 devices I’ve configured+downgraded in 1 step.
    (Where did I ever stated such thing?)
  • I do not use ANY port or IP related thing on the device, everything is turned OFF. (Except syslog)

I use a trivial setup:

  • The analog Landline is connected to the LINE input,
  • the analog phone is connected to the PHONE port
  • and the LAN (in bridge mode) is connected to the router to send syslog info to the server.
    That’s it. No IP phoning used at all.

If someone is calling/ringing >> the CID will be sent to the syslog server
If someone is trying to dial out >> the HT503 should simply let doing it without interfering.

Obviously I can use an “Y” adapter to connect the 2 devices parallel, but in this case it can happen that the CID has no time to “come through” if someone picks up the phone too early.


#5

The {SOLUTION} was to:

1.) access the web interface of the HT503 => Advanced settings =>
set: “Disable Voice Prompt: NO” ( click on APPLY)

2.) Than I had two options to dial out:

++ the first: pick up the handset connected to FXS port and dial: * 0 0 (star,zero,zero)
then you will hear the PSTN line dial tone, then dial the external number

++ the second is to access the FXS PORT page and to change the current dial plan to this one:
( delete the old one and add this one):
{ L: 06x+ | x+ | *x+ | xxx+ }
( click on APPLY)

Tested with newest Firmware: 1.0.16.3 beta.

Many Thanks for Yassir (Grandstream’s helpdesk employee) :slight_smile: