14 Jul
2011
14 Jul
'11
3:40 p.m.
Am 13.07.2011 10:07, schrieb MingHon:
Hi,
i tried set canreinvite=yes in asterisk but kamailio/rtpproxy still trying to send rtp traffic to asterisk.
That should not happen. You have to investigate why. You have to take a look at the SIP signaling during and after call setup.
You should see reINVITE messages from Asterisk to the clients. Take a look at the SDPs in those requests and their responses to find out if they are malformed.
regards Klaus