[asterisk-users] rfc2833 vs. sipinfo and network weirdness

[Michael]

Further to a recent post about a problem whereby the server continues to spew 
packets to the phone after hangup (sometimes, not every time), I have found 
that this problem appears to be alleviated by using RFC2833 instead of SIP 
INFO, however in switching to RFC2833 I introduce another problem - that DTMF 
tones for navigating menus become unreliable.

RFC2833 and SIP INFO are the only 2 options supported by the phone.

Inbetween the phone and the server is the internet and 1 NAT traversal. 

Any ideas?

Michael