In my setup, which is FreeBSD, using pjsip 2.5.5 as sip backend I am observing a regression when testing the latest Release Candidate.
Any calls get refused and the following error shown on console: [Nov 22 10:49:26] WARNING[101105]: res_rtp_asterisk.c:2400 int create_new_socket(const char *, int): Unable to allocate RTP socket: Protocol not supported [Nov 22 10:49:26] WARNING[101105]: res_rtp_asterisk.c:2665 int ast_rtp_new(struct ast_rtp_instance *, struct ast_sched_context *, struct ast_sockaddr *, void *): Failed to create a new socket for RTP instance '0x805647c30' [Nov 22 10:49:26] ERROR[101105]: res_pjsip_sdp_rtp.c:184 int create_rtp(struct ast_sip_session *, struct ast_sip_session_media *): Unable to create RTP instance using RTP engine 'asterisk' Please note that with asterisk 13.12.2 everything works fine. No other change was made to the system. Is this a known issue being worked on? Should I file a bug report? I am the maintainer of the FreeBSD Asterisk port in the ports tree, and I routinely test Release candidates when available to speed up updating the port. So I'm especially interested in this issue being investigated, otherwise I'll have to hold up updating the FreeBSD Asterisk port. Thanks in advance. -- Guido Falsi <m...@madpilot.net> -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
