Nothing is wrong here, it's just that the OPTION method is not handled by
SIP Communicator. There're also various methods not handled by the SIP
Is there a way to prevent Asterisk from sending those methods? (maybe in
sip.conf?) Any ideas?
I am curious, did the RTP stream work for you on Asterisk?
With me it worked. As my endpoints are in the same LAN as Asterisk I had to
take out the STUN server as this caused the connection to go wrong.