Am 28.11.2012 20:58, schrieb Konstantin M.:
Jeremy, it is doesn't work at all. I've made a lot of changes to that patched asterisk to make it working and no luck. However, ast11 has fully supported webrtc, but I heard no voice during a call.
Same here. I also tried the Doubango patch but it doesn't help. regards Klaus _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
