Digital receptionist not accepting key in IVR from outside calls

[jabir]

I have setup Digital receptionist to accept all calls. It plays an IVR and in order to get to a certain extension you have to press a key. this works fine if i call from 3cx app but when i try to call from my phone, i can listen to the IVR but when i press key nothing happens. It seems that Digital receptionist is not receiving those key strokes when calling from my cell phone.
I have a PSTN line connected to cisco router and that cisco router has a SIP trunk to 3cx server.
I am using 3cx version 16, which is installed on our server (Windows server 2019) in our office.
Any help in figuring this out would be great, thanks in advance.
below is the screen shot of my Digital receptionist setup:

 

[craigreilly]

"when you cal from your phone" - assume you mean a cell phone or something into the PSTN number. Have you tried another external device? If it too is having same issue - I would check the config of the Cisco and how it is sending the tones.

 

[jabir]

"when you cal from your phone" - assume you mean a cell phone or something into the PSTN number. Have you tried another external device? If it too is having same issue - I would check the config of the Cisco and how it is sending the tones.

that is correct i am calling my PSTN number which is connected to cisco router through my cell phone. i tried calling from different cell phones but same issue. Below is the basic configuration of router concerning the calls to 3cx (I have masked the actual number with ###)

voice-port 1/0/0
no battery-reversal
input gain 10
output attenuation 10
no comfort-noise
timing hookflash-out 50
connection plar opx 2222
description trunk ##########
caller-id enable

dial-peer voice 1111 pots
destination-pattern 2222
incoming called-number .
port 1/0/0



dial-peer voice 1 voip
preference 1
destination-pattern 2222
voice-class codec 1
session protocol sipv2
session target ipv4:xxxxxxxxxxxxx
no vad


dial-peer voice 3333 pots
destination-pattern .
port 1/0/0


dial-peer voice 2 voip
description **Outgoing Call to SIP Trunk**
destination-pattern .
voice-class codec 1
session protocol sipv2
no vad

 

[leejor]

It may be, the DTMF Method, set on the PSTN Gateway, or, if using Audio DTMF, the Codec. DTMF audio requires that you not use a low bit rate Codec,or tone will be distorted enough to fail.. I'm assuming that internal calls, to the IVR, work as intended. If you call an extension from an outside line, then press some numbers, do you hear them?

 

[jabir]

It may be, the DTMF Method, set on the PSTN Gateway, or, if using Audio DTMF, the Codec. DTMF audio requires that you not use a low bit rate Codec,or tone will be distorted enough to fail.. I'm assuming that internal calls, to the IVR, work as intended. If you call an extension from an outside line, then press some numbers, do you hear them?

Internal calls works fine. is there any way to make it work like when i press a key on my cell phone during a call, the IVR should divert my call to intended extension.

 

[craigreilly]

check the pstn gateway DTMF method.

 

[jabir]

check the pstn gateway DTMF method.

I am not that much technical. i added "dtmf-relay sip-kpml rtp-nte" command to both voip dial-peers but it doesnt work. Can you guys guide me in right direction.