Hi, while testing current release candidate 1.8.0-rc2 I stumbled on a weird behavior. I did not find any hints in the archives or at the bug tracker.
Two SIP-Clients are connected (both on the local net, no NAT). The RTP stream flows directly between the phones. If I set phone A on hold, the music on hold is played. On the CLI I see the following message running: WARNING[2470]: res_rtp_asterisk.c:1939 bridge_p2p_rtp_write: RTP Transmission error of packet to (null): Invalid argument The message is running until the phones are connected again. In the meantime the CLI is nearly unusable. This does not happen, if I configure asterisk to stay in the media path. Is this a new bug or do I something wrong? File sip.conf looks like this: [general] bindaddr = 0.0.0.0 disallow = all allow = alaw allow = ulaw language = de allowguest = no fromdomain = 192.168.10.70 tos_sip = 96 tos_audio = 184 [katrin] type = friend host = dynamic callerid = Katrin Wemheuer <200> context = Standard mailbox = 200 [max] type = friend host = dynamic callerid = Max Müller <245> context = Standard mailbox = 245 Thanks, Karsten -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
