Never mind, it seems the issue was triggered by my soft client not using STUN
Fixed that bit now :) Mark. On Fri, 7 Jun 2019 at 10:53, Mark Farmer <[email protected]> wrote: > Sorry - version info: > > opensips -V > version: opensips 2.4.5 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll, sigio_rt, select. > git revision: d025b4f61 > main.c compiled on 11:39:27 Apr 12 2019 with gcc 7 > > > On Fri, 7 Jun 2019 at 10:45, Mark Farmer <[email protected]> wrote: > >> Hi everyone, I'm trying to solve an issue related to reception of early >> media. I am getting 183 messages back to establish the audio path but >> OpenSIPS is passing the 183 back to an Asterisk box without changing the >> SDP "c" parameter. So Asterisk tries to send audio direct instead of to the >> rtpproxy. >> >> Doing some debugging, I can see that my reply route doesn't seem to be >> matching the 183 because I never get any logs from the reply route. Can >> anyone see anything wrong with it? I've checked the routing script & the >> reply route seems to be armed. >> >> onreply_route[DROUTING] { >> >> if (is_method("BYE|CANCEL")) { >> sip_trace("htid","d"); >> rtpproxy_unforce(); >> } >> >> #if ( $rs >= 200 ) >> #$acc_extra(to_usr) = $tU; >> >> if ($rs=~"(2[0-9][0-9])|(183)" && has_body("application/sdp")) { >> xlog("Processing reply $fU"); >> if (is_from_gw("1")) { >> xlog("Reply from Asterisk PBX"); >> setflag(INT_R); >> } else if (is_from_gw("2")) { >> xlog("Reply from Provider"); >> setflag(EXT_R); >> } >> } >> >> if (isflagset(INT_R)) { >> remove_hf("P-Asserted-Identity"); >> rtpproxy_answer("corwfei"); >> } else if (isflagset(EXT_R)) { >> rtpproxy_answer("corwfie"); >> } >> } >> >> >> Many thanks for any & all help. >> Mark. >> >> > > -- > Mark Farmer > [email protected] > -- Mark Farmer [email protected]
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