u send your calls through the asterisk anyway).
You might also consider using the RTPProxy with the patch in the
sip-router-repository. With the patch, the RTPProxy will trigger a
teardown of calls (via XML-RPC) if the RTP-Session has a timeout.
Carsten
2011/6/23 Brett Woollum :
> Hi
reet NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Jun 23, 2011, at 1:10 AM, Brett Woollum < br...@woollum.com > wrote:
Hello,
We are running Kamailio as a registration point for our SIP phones, which then
interac
Hello,
We are running Kamailio as a registration point for our SIP phones, which then
interacts with Asterisk. SIP registrations are processed by Kamailio, but
everything else is passed to Asterisk. The Kamailio configuration is close to
the article at:
http://kb.asipto.com/asterisk:realtime