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AT&T Conferencing and DTMF Issues

I am looking for some feedback from voice users who had SIP trunk to PSTN and dealed with AT&T conference system. I have two issues here:

1. DTMF is not recognized by AT&T conference  number (+1 215-446-0183). I have set the UCM 8.0 SIP trunk using RFC2833 as DTMF method. I read a doco saying Cisco rtp-nte DTMF is not so compatible with AT&T thus I have to add these  config for inbound dial-peer to UCM.

rtp payload-type nse 99

rtp payload-type nte 100

Has anyone used the above and has got the DTMF work? Is there anything else I need to setup for getting away of the DTMF problem?

2. the AT&T  conference calls were always dropped at around 9 - 10 minutes. I tried with the below config but still having call drop:

voice service voip
  sip-profiles 100

voice class sip-profiles 100
request INVITE sip-header Allow-Header modify " UPDATE, " " "
request REINVITE sip-header Allow-Header modify " UPDATE, " " "
response 200 sip-header Allow-Header modify " UPDATE, " " "
response 180 sip-header Allow-Header modify " UPDATE, " " "

Appreciate any info for fixing the At&T conference and DTMF problems above.



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