Hello, I am runing asterisk on a embedded linux and am having some RTP audio issues at the beginning of the call: the comfort noise packet seems to be opening the pinhole in the firewall though I don't understand why it is not already opened. Then audio is then transferred correctly between caller and callee through the asterisk bridge. The SIP INVITE is received on a WAN interface and then I dial out to another SIP channel through the same interface. CLI output with RTP debug shows that Packet2Packet is only started and RTP is only sent by asterisk after the first rtpkeepalive timeout. If I sniff at a mirroring port in the network I can see the first RTP packet going from my caller to the asterisk server yet it seems that it is never received (or it never reaches) asterisk (it is a direct route). All firewall rules on the asterisk box are setup for the range of ports defined by rtp.conf (10k-11k in mycase); that is consistent with the SDP signaling generated by asterisk for the INVITE OUT and for the 200 OK back to the caller in the media description attribute. Watching iptables live activation does not show any RTP packet blocked at the beginning of the call. netstat shows: netstat -an | grep udp | grep 10 netstat: no support for 'AF INET6 (tcp)' on this system netstat: no support for 'AF INET6 (udp)' on this system netstat: no support for 'AF INET6 (raw)' on this system udp 0 0 216.54.141.148:10554 0.0.0.0:* udp 0 0 216.54.141.148:10555 0.0.0.0:* udp 0 0 216.54.141.148:10102 0.0.0.0:* udp 0 0 216.54.141.148:10103 0.0.0.0:* as I am using bindaddr=0.0.0.0 in the sip.conf. I have multiple NICs on that box, could it be a problem or ...? Thanks for any suggestion, Sebastien. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081001/35a3f204/attachment.htm