DTMF not collected on vector using SIP Trunk CM 6.3

[AvayaIT]

Hello,

i have strange issue with my AVAYA CM 6.3 , on incoming calls to the vector using SIP Trunk , the caller hear the DTMF tones and no digits are collected (the vector is used to collect digits and route to extension based of the collected digits)

anyone can suggest a solution or where is the issue coming from ?

Best Regards

 

[Sean655]

You'll have to work with the carrier most likely. I don't know of any tweaks you can do on the pbx, I've always had to work with the carrier on things like that, even on TDM circuits.

 

[AvayaIT]

thank you for your reply , i will check with the carrier

 

[kyle555]

traceSM. Look at the media types in the SDP in the invite.

Probably something like RTP/AVP 0 97

the 0 means G711
The 97 or 101 or whatever means RFC2833 DTMF. So, it's in the RTP stream, but it's not an audible tone you can hear, you literally see in wireshark a DTMF Event.

If the SDP only had media type 0, then that's only G711 and that means the audio is only coming in-band. For you, that sucks because you'd have to configure your sig group for that DTMF type and not listen for RFC2833 DTMFs. And, you need to burn a DSP for the length of every call. Suppose you have 1000 trunks and 1000 sets and a G450 with 320 DSP. A vector for an autoattendant could connect 1000 trunks with 1000 sets very easily so long as they all didn't call at once. Well, if you need in-band DTMF, you need 1000 DSPs to listen all the time.

Let's pretend your carrier peers with a cableco to access each other's PSTNs. Let's say that little rural cableco has some really old cable modems with SIP ATAs in there that don't to RFC2833 DTMF. Nobody is listening to that little cable modem's stream from the subscriber's house to the cableco to the SBC for your provider to your PBX to see if there's an audible DTMF in there and changing it to a RFC 2833 tone for your sake.

Avaya experience portal can listen for both at the same time. And I've always thought it'd be cool if CM could have a direct audio sig group unshuffle and use a DSP to listen for in-band tones if it saw a voice call with just media type 0, but alas, I don't think it's meant to be.

 

[jimbojimbo]

As Kyle555 noted look for the telephony event type. On the SIP Trunk form on page 4 you can match it. If blank it is 127 by default. Typically seeing 101 (NTE).