[asterisk-users] rtptimeout on Asterisk 1.4.x

[Rodrigo P. Telles]

Hi Folks,

Since I upgraded my asterisk box from 1.2.x to 1.4.x (1.4.10.1 now) I noticed some dead calls "apparently" running for
more than 8 hours.
I'm using rtptimeout=60 and rtpholdtimeout=120 and found some log messages like this:

chan_sip.c: 'SIP/XXX-085a9308' will not be disconnected in 61 seconds because it is directly bridged to another RTP stream

I can kill that calls using 'soft hangup <channel>' but I'd like to know if its a new BUG introduced in 1.4.x releases
and if possible, how to fix this?

Thanks in advance.
Rodrigo P. Telles