HT503 two pstn issues



I have HT503 connected to huawai ft3110 as pstn line. FT3110 is a “Fixed wireless terminal”.
FT3110 works fine with any phone I have attached to it (phone dials every time, rings long for incoming calls and shows caller ID).
Currently I faced two issues.
The first issue is: If I dial *00 I can hear the dial tone but the DTMF tones always dial a wrong number. But if I route some numbers to pstn (L: x… in FXS tab), then these numbers are dialed absolutely correct. The difference is that I don’t hear the dial tone. I just dial the number and ht503 somehow sends it to the pstn correctly. I tried different AC and Gain settings but I always get a wrong number response (sometimes it is wrong code). Any of my phones can dial when is attached to ft3110. Any ideas how to solve this?

btw: voip to pstn works well and dials correctly.

The second issue is even more interesting. The pstn incoming call starts ringing the phone on FXS port but after two rings the phone stops ringing. I dial it from my mobile phone and I hear that it continues to ring. In practice phone on FXS port give up after the second ring and after the caller ID is detected and shown on the phone display.
I found that the above behaviour is related to the caller ID. If I change the "FSK Caller ID minimum RX Level " to -70db the phone rings all the time but doesn’t show a caller ID. If the caller ID is shown (at default level -40dB) then the phone on FXS port stops ringing. Any ideas about it?

“Ring Timeout” is default value
“Number of Rings” is set to 30

I know It is long but I spent some time on it:)


The wrong dial issue is mostly likely related to the DTMF tones not being synchronized correctly between the two devices.

I assume the connection is:

  GSM<---->FT <-----(fxo)-------->HT<----(fxs)-------> Phone

I am not certain I understand the need for the HT. You indicated that -“FT3110 works fine with any phone I have attached to it”. The HT also has an IP interface for SIP and it is not clear to me where this is coming into play other than -“voip to pstn works well”. What is the VoIP connection? More info is needed to understand just how set-up connection wise.

By dialing the *00, this allows the HT to basically connect the FXO and FXS lines together in order to present the second dial tone. Once this tone is heard, then you can input the desired dial string. The DTMF tones in this case are then transmitted inband with the audio from the phone to the FT. There are DTMF channel settings in the FXO side that allow you to set the duration and timing of the tones. There is also a setting for two-stage dialing, which may need to be engaged if one stage dialing is currently enabled.

There are a number of settings used in the HT to control how and when a call is delivered.
It really has nothing to do with the gain per se, but you can influence it.

The number of rings is designed to provide enough time for the HT to capture the CallerID (CID) on the inbound call so that it can relay same. In the U.S., for example, the CID is delivered between the 1st and 2nd ring so the setting would normally be set to 2. It is always assumed that once the CID has been captured that there is no reason to delay the delivery of the call any further as the caller may give up waiting. By adjusting the FSK Rx level, you are preventing the phone from being able to detect the CID. You should leave the detect at the default level. In the FXS side,there is also a ring timeout. This needs to be set if not already.

Not knowing how the VoIP comes into play, I assume that what might be happening is that the forward to the VoIP side is answering the call instead of the analog phone and thereby canceling the FXS phone side need to ring further.