2 Jul
2012
2 Jul
'12
10:50 a.m.
Hello Just switched this weekend from asterisk to freeswitch on freebsd 9.0. And so far I was able to implement basic dialplan for inbound/outbound calls (o; What I'm trying to find is a similar configuration option from asterisk sip.conf where individual user agents can have SIP reinvite disbaled with "canreinvite=no". Fomr the Freeswitch documentation it states that all RTP streams are passed through by default, but I still see SIP clients trying to send UDP streams to my box directly. Also setting "<param name="inbound-proxy-media" value="true"/>" in sofia.conf.xml doesn't solve this problem. I just want the UDP streams frmo outside are only allowed from my SIP trunk provider. thanx in advance richard