Am Mittwoch, den 06.10.2010, 15:11 +0200 schrieb Karsten Wemheuer: > 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.
for the archives: This behavior seems to be fixed in 1.8.0-rc3. 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
