22 Jun
2012
22 Jun
'12
11:14 a.m.
Hi Shaun!
Your problem description is too short to give you any good help.
Use tcpdump (or other tools) to capture the scenario with Asterisk and
Kamailio. Then compare them to find out why it doesn't work.
Is media sent directly to Asterisk then it ca not be the problem of
Kamailio.
I hope the mobile client is smart enough to also send a reINVITE when
getting the new IP address (of the mobile connection) with proper
Contact header - otherwise it can not receive SIP requests from Asterisk.
regards
Klaus
On 20.06.2012 18:07, Shaun Clark wrote:
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!
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