20 Jun
2012
20 Jun
'12
12:07 p.m.
The use case is that I have a SIP client registered to Kamailio talking to
an Asterisk box connected to the PSTN. The client is a mobile phone and the
user is connected to wifi. The user then steps out of wifi range and the
phone drops the connection and picks up the 3g data connection. I want the
media stream to reconnect to the client and the call to resume without
having to redial. This works now if the client is directly connected to the
Asterisk machine, but not when I am routing through my Kamailio server. How
do I go about this, examples are always appreciated, thanks!