AS5400XM DTMF problem

alex goshtaei · ‎05-20-2009

Hi All,

AS5400 router is connected to 10 T1 from TDM side and SIP trunk to the IVR. It doesn't recognize DTMF tone when I call from PSTN to the IP. in the IP site is connected to IVR system.

this is my config:

dial-peer voice 100 voip

destination-pattern 281[2-9]......

progress_ind alert enable 8

progress_ind connect enable 8

session protocol sipv2

session target ipv4:10.1.1.1

incoming called-number .%

dtmf-relay rtp-nte cisco-rtp

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad

!

dial-peer voice 7 pots

description ** Incoming call from PRI **

max-conn 100000

incoming called-number 281[2-9]......

no digit-strip

direct-inward-dial

port 4/6:D

any suggestion would be very appreciated.

thanks

Alex

paolo bevilacqua · ‎05-20-2009

Try removing dtmf-relay and set codec to g.711.

The IVR may like that better, and voice quality will improve.

alex goshtaei · ‎05-20-2009

thanks for reply,

actually I did before and there was no luck.

paolo bevilacqua · ‎05-20-2009

What the IVR vendor says, which dtmf relay do they support ?

alex goshtaei · ‎05-20-2009

it's VoIPswitch and it support RFC 2833.

paolo bevilacqua · ‎05-20-2009

You can take a debug voip rtp dtm-relay to see how the messages are sent and give the ball to the IVR vendor.

Nicholas Matthews · ‎05-21-2009

Hi Alex,

A few common things:

debug voip ccapi inout and make sure you're matching an incoming and outgoing dial peer correctly. You can search for peer= to find them (It's right under the big block ccapi creates).

You can also remove cisco-rtp, since that's proprietary and nobody uses it.

There have been some interop issues with the way we do RFC 2833 by default, you may want to add "dtmf-interworking" to the dial peer to see if that changes things.

Another common problem is the other end will advertise a payload type for 2833 that we use for something else, like 100 with NSEs. You can look at the 200 OK they are sending back and see if they're sending 101 in the m-line or not. You should see a a=fmtp 101 1-16 or something similar. If that number is 100 for example, you can do a 'show voice dial-peer 100' and it will list the static payloads for all the codec types. Then change it with the 'rtp payload-type '.

And actually the debug Paolo mentioned is broken (for some call flows) and isn't suggested. 'debug voip rtp session named' is preferred.

hth,

nick