3CX with Skype Gateway - not passing keytone

[ITSUPP]

Hi,

I have a working 3CX version 11 SP2 with Skype Gateway. I can make an outbound call (not doing inbound) via Skype gateway and I can hear the voice both ends. A little problem is when it connected, if I press any number, it doesn't passed on to the receiver/destination.

For example if I call to any number with automatic system which requires to press 1 to talk to A or press 2 to customer service.
Currently, once connected, when I pressed 2, the receiver/destination end didn't get my entry. From the source/caller, I could hear a tone when I pressed 2.

Thanks in advance for the help and advice.

 

[markshehan]

It has been a while since I played with Skype in this way, so bear with me

Can you make the same call through Skype and press the same buttons and make it work?

Also, could you take a wireshark capture and post it and we can see the DTMF that is being sent (and maybe received too) so we can try and diagnose it for you

 

[leejor]

My understanding of the workings of the Skype gateway in 3CX, is that it relies on the audio "bits" of the computer you are running it on. That infers that DTMF is passed as audio and not as a SIP message. So...it may have something to do with the DTMF settings on the phone you are calling from, or how well Skype is able to interpret the length and level of the tone you are sending. There is probably a way to monitor/confirm this by "tapping" into the audio card/circuits on the PC.

 

[ITSUPP]

Hi, sorry for the late reply and thanks for the input.

If I use Skype application, to the same number, yes I can access the options available.
Just not via 3CX.

I installed Wireshark then made a call, the log in wireshark:
Source Destination Protocol Length Info
----------------------------------------------------------------------
my_public_ip 125.213.184.18 UDP 71 Source port: 18340 Dest port: 12340 (this line got error)
125.213.184.18 my_public_ip UDP 91 Source port:12340 Dest port: 18340 (no error)

and keep showing those above while the call connected, with or without my inputs.

The error (in detailed field) shows:
Header Checksum 0x0000 [incorrect, should be 0x90f0 (maybe caused by "IP checksum offload"?]
[Bad: True]
[Expert Info (Error/checksum): Bad checksum
The Source and Destination GeoIP are both: unkown

I am not sure whether those are the caused by the entry.

I use the softphone to do the testing. The softphone is installed on the 3CX server. I also tried installed the apps on my iPhone and the result are the same.

Thanks.

 

[markshehan]

Hi, sorry for the late reply and thanks for the input.

If I use Skype application, to the same number, yes I can access the options available.
Just not via 3CX.

I installed Wireshark then made a call, the log in wireshark:
Source Destination Protocol Length Info
----------------------------------------------------------------------
my_public_ip 125.213.184.18 UDP 71 Source port: 18340 Dest port: 12340 (this line got error)
125.213.184.18 my_public_ip UDP 91 Source port:12340 Dest port: 18340 (no error)

and keep showing those above while the call connected, with or without my inputs.

The error (in detailed field) shows:
Header Checksum 0x0000 [incorrect, should be 0x90f0 (maybe caused by "IP checksum offload"?]
[Bad: True]
[Expert Info (Error/checksum): Bad checksum
The Source and Destination GeoIP are both: unkown

I am not sure whether those are the caused by the entry.

I use the softphone to do the testing. The softphone is installed on the 3CX server. I also tried installed the apps on my iPhone and the result are the same.

Thanks.

Dont worry about the wireshark message - that is pretty standard and is just a checksum error not a problem with what you are sending.

Your problem is, as Leejor said, with the way you are sending the key presses (DTMF) for the menu options. There are two ways to do this with SIP, either as beeps in the audio or as SIP messages. Skype only handles the beeps. But your phone is configured to send the sip messages as it is a sip phone. Most can be changed to send the beeps. But it is a phone setting not a 3cx settings. 3cx just detects what it gets and passes it through (as it can handle both)

 

[leejor]

Many devices can be set to send both audio and a SIP message, in most cases, it will work. In rare cases (certain trunks/gateways) it causes "double digits".

One other issue that arises, with many devices that can send audio DTMF (most/many electronic phones), is that the tone length is pre-set by the device no matter how long the key is pressed, and in some cases, this is too short (and too low ?), to be useful (recognized), at the far end. This is why DTMF SIP messaging, and re-generation, in some cases, is generally used.

As Skype (the original version) was conceived as a computer to computer service, DTMF transport was probably not given a lot of thought.

 

[ITSUPP]

Hi,

Somehow somewhat I quite disagree that the problem is with the device.
As a newbie, sorry for that, as I am not the expert in VoIP and still learning bits by bits. :S

Further to my investigation and testing (that's why I am disappear), I tested using SipToSis (previously, SIPTheeSkype). With the same setting on the softphone (connect to 3CX PBX -> 3CX Skype Gateway vs connect to SipToSs directly), except the SIP port, the SipToSis passed on the DTMF signal.
So I have a feeling that somehow the 3CX PBX not passing on the DTMF signal to 3CX SkypeGateway or the 3CX SkypeGateway that is not handling DTMF?

I prefer to use 3CX PBX as the server that bridge to the Skype Gateway and I think with 3CX it's a bit clearer in the voice quality.
My guess that the 3CX SkypeGateway won't be developed anymore (improved to newer version). Or still there is a hope here?

The only little minus with SipToSis is that before the call is answered by the callee, SipToSis somehow send the signal to the caller as it is answered. Not good for billing. This doesn't happen using 3CX SkypeGateway.

Thanks.

 

[leejor]

Further to my investigation and testing (that's why I am disappear), I tested using SipToSis (previously, SIPTheeSkype). With the same setting on the softphone (connect to 3CX PBX -> 3CX Skype Gateway vs connect to SipToSs directly), except the SIP port, the SipToSis passed on the DTMF signal.
So I have a feeling that somehow the 3CX PBX not passing on the DTMF signal to 3CX SkypeGateway or the 3CX SkypeGateway that is not handling DTMF?

I prefer to use 3CX PBX as the server that bridge to the Skype Gateway and I think with 3CX it's a bit clearer in the voice quality.
My guess that the 3CX SkypeGateway won't be developed anymore (improved to newer version). Or still there is a hope here?

I'm not sure how SipToSis actually works, if it uses DTMF SIP messages, or DTMF audio. With the Skype gateway, I believe that any DTMF that is to be passed on to Skype has to be audio as I don't believe that 3CX has any way of converting SIP messages to DTMF, that is the job of a gateway (PSTN lines). Which means that the set you are using has to generate DTMF audio, and it has to be at a level and length that will make it over the Skype circuit. What the Skype circuit does with it after that....well

As far as another version of the 3CX Skype Gateway, I would not hold my breath, It could happen, but don't count on it. I think it's just been left to die a slow death.