I tried once again to switch an asterisk install to use pj instead of chan_sip, and am seeing an odd error.
This particular one is dealing with unwanted calls (endless sewer style of reply), and given this sdp: v=0 o=200107170215185:5060 16264 18299 IN IP4 0.0.0.0 s=pplsip c=IN IP4 0.0.0.0 t=0 0 m=audio 25282 RTP/AVP 100 6 0 8 3 18 5 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 pj returns PJMEDIA_SDP_EMISSINGRTPMAP, which means it cannot find an rtpmap line. Even though two exist. I see that example has 0.0.0.0 for the o and c lines, so obviously the attacker doesn't want audio. But I'd still like to waste some of their cycles dealing with a series of 180s. Chan_sip would send the call to the dialplan. Should pj not also? -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
