[asterisk-users] Channels hanging when SIP phone gets resetduring call

[Olle E Johansson]

22 feb 2007 kl. 12.20 skrev Steve Langstaff:

> Are the RTP timers applicable with canreinvite=yes ?

how could we possibly check RTP if the RTP doesn't touch or network  
card at all?

The timers are only used when we have RTP streams going to us. If the  
RTP stream
is redirected, it's up to the end points to hangup due to media failure.

The way to solve this is to implement the SIP timer extension.

/O