Hi, I have configured the 2 level IVR. First level is Welcome Message and Second level is for Language selection. IVR perfectly works fine from the GSM mobile devices and PBX connected PSTN devices however the issue seems to be from the PSTN directly connected to home/offices. IVR menu option doesn’t recognized when pressed from these devices and repeat the Welcome Message time and again.
Service provider informed me to check the DTMF and codec and below is the dtmf setting
dtmfmode=rfc2833
codec
allow=ulaw,alaw,g729
I am not getting the concrete solution as service provider tells to fix this issue from our side. They have also suggested to maintain the power gain from 0-25 db but I couldn’t find those setup in asterisk system as it is available in PBX box device.
I’m a bit confused because you appear to describe everything as being PSTN, but then suddenly throw in dtmfmode, which only applies to VoIP. My best guess is that your main access to the PSTN is via SIP, but you have some local analogue phones that don’t go through the SIP provider, and you have some analogue devices that go out to PSTN, and then come back in through the SIP provider. I’m not clear about the GSM devices. Are they coming in through the SIP provider, or through a local GSM gateway. Also, I’m not clear as to whether the locally connected analogue devices are going through an ATA or direct to an FXS ports on a card plugged into the Asterisk PC>
I would note that G.729 will not support in band DTMF reliably, so the first thing I’d want to do was to check whether the provider really is using RFC 4733/2833. The most reliable way of that is probably to enable RTP debugging (rtp set debug on, I think), and look for the special packet types used to carry DMTF events.
Basically what I want to say is we have sip configured and whenever the PSTN user dials to the SIP DID number the IVR menu doesn’t passed to 2nd level…it time and again repeats the Welcome Message and no DTMF signal passed from the PSTN device.
Use RTP debugging to see if you are actually getting the DTMF sent as telephony events. If not, use the protocol logging option for your SIP channel driver to make sure that telephony events is being offered by both sides in the SDP. If missing from the ITSP side, it is their fault. If it is present on your side, it is their fault. Otherwise check you put the dtmfmode in the right place and there are no error messages about it when Asterisk is starting.
With my experience, two problem can occur in this case:
If you see DTMF in packet capture so Voice gateway already send DTMF to Asterisk. Problem is DTMF mode mismatch between asterisk and voip gateway. You change it and everything will ok.
If you don’t see DTMF signal (in packet capture file) send from voice gateway to Asterisk, so may be voice gateway is not config correctly with local tone, you should reconfig it, tone is different between other telephone provider.