[Asterisk-Users] Polycom config and DTMF problems

[Douglas E. Warner]

On Wednesday 05 October 2005 08:02, Douglas E. Warner wrote:
> I'm going to try to capture the RTP stream and
> see if it's being sent inband,

So, I've captured the RTP stream and played it back (nothing but payload type 
0 packets, btw), and there was no DTMF in there at all - I held down the zero 
key a lot of the time, and talked into the handset the rest of the time; the 
time where I was talking could be heard, but there was nothing during the 
times when I was holding down the zero key.
Could this be a firmware bug?  Is anyone else having problems w/ Sip 1.5.3?  
Defective phone?

-Doug

-- 
Douglas E. Warner    <dwarner at ctinetworks.com>     Network Engineer
CTI Networks, Inc.   http://www.ctinetworks.com    +1 717 975 9000
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