DTMF input not being applied

[cknott2243]

I am using v10 with the call center modules, up to SP5. This is connected to PSTN lines using a grandstream 4108.

I have been receiving reports randomly from customers and internal people saying when reaching the receptionist it does not apply their DTMF input, for example, if trying to do direct extension it does not allow it and acts like no keys were pushed. In other cases they say they are just trying to press an option within the menu.

I cannot find anything in the logs so far or in my own tests that show a problem. I know the obvious cause is probably a non compatible DMTF setting but this would seem to be odd for the normal caller. Maybe an issue with the 4108 but I would think it would be all or nothing if so, not let some work and others not.

Anyone have any ideas? I am new to 3CX and just went live so could just be me not knowing something simple in the system.

Thanks.

 

[leejor]

Intermittent troubles are the worst type to work with as you have to establish some sort of pattern before you can point your finger at something.

DTMF will can use several methods, either audio, which doesn't function well unless you are using the G711 Codecs, or, one of several SIP message types. With the SIP messages, you need to be sure that everything in your network is set to accept the same DTMF "method". Sometime sets will default to "auto", which will work in most cases. but. can cause problems occasionally.

The problem with DTMF audio can be level, and/or duration. Some sets will only send a very sort DTMF burst (many cordless and mobile phones), and if the incoming level (external callers) is a bit low, the tones just don't register all the time.

If you can find some sort of pattern, who is having the issue (type of external caller, long distance/local/mobile, certain internal sets), it might help in tracking down a solution.

 

[RLester]

I had the GXW4108 and had the same trouble. I had to change my phones and the gateway to use SIP-Info (only) I think this is option 4 in the gateway. I also turned up the gain to send a stronger signal.

 

[cknott2243]

Thanks everyone. I am still trying to find a model where this can be duplicated.

Rlester - Can you give me a little more detail on where to find this option in the 4108 and how to turn up the gain? to receive a stronger signal?

Thanks again.

 

[RLester]

Yes I can but I am sending this back and right now just today testing a Patton 4114 (so far grate!! no BS with this one).

I am out of the office now but I have pictures of some of the config screens on my phone.

I think on the channels page. "Channel Voice Setting" The Tx to PSTN I sat to ch1-8:10;

And just below that is "Channel Specific Setting" DTMF Methods I did Ch1-8:4;

 

[leejor]

When you start changing the gain settings you run the risk of echo problems. Incoming gain increase, can = outside callers hearing echo. Do small changes and make note of the original setting of anything you change.

 

[RLester]

I shoot high with 12 at first and did have to move it down for just that reason. I think it was just trying the extremes and I just never had a need to change it most likely I could adjust it down some.

On a new note. Are you using 1 stage dialing? How many lines do you have?

 

[cknott2243]

Rlester - Thanks for the info. I have changed those settings and will monitor. I am really not sure about the one stage dialing as I'm new to this whole setup of a IP-PBX.

I will give it a few days and see if there are other reports.

Thank You.

 

[RLester]

Did you change the DTMF method in your phones? Have you called some place you’re self and tested? I call a place and it will read me the numbers that I dial back to me, maybe you can find some place that you deal with that would act the same for you. (Account verification)

1 Stage Dialing is where you pickup the phone and just dial your number that you want to call. (NOT dialing something like 9 to get an out side line)

 

[leejor]

1 Stage Dialing is where you pickup the phone and just dial your number that you want to call. (NOT dialing something like 9 to get an out side line)

That's not entirely true, as one stage dialling is the setting in gateways that the majority of users will use with 3CX, and can be used whether you dial 9 (an access digit) or not. In fact, many SIP phones and ATA's can be programmed to simulate cut through to outside dialtone, you would have no idea that it wasn't actually happening.

Two stage dialling, can cut you through to actual PSTN dialtone , in some cases, in most, it will cut you to Gateway generated dialtone and then the gateway will use it's own internal dialplan to screen numbers before passing them onto one of the phone lines. Two stage dialling "disables" any sort of number restriction , screening or automatic routing within 3CX and passes that function onto the gateway.

Of course this all depends on the features available in your particular gateway.

Two stage dialling is handy, if you register the gateway with a VoIP provider, rather than as a trunk with 3CX. You could then call one of the VoIP numbers, get dialtone from the gateway, and dial a number out on one of the PSTN lines. The SPA3102 is very good at this as it has a couple of security features, PIN and/or caller Id screening.

 

[RLester]

I see leejor thank you for the explanation. I was not thinking about dialing a particular digit to use for routing in rules to access different set of trunks. I’m not sure if I get all of what you’re saying as I am very new to this.