On Mon, Feb 7, 2011 at 6:48 PM, Henk Hesselink <opensips-us...@voipro.nl> wrote: > Hi Chris, > > That config should't touch the Contact header, and yet that's also been > modified: > > In: Contact:<sip:+13038382386@208.94.157.10 ... > Out: Contact:<sip:+13038382386@67.212.153.178 ... > > Are you sure nothing else is touching the message? > > Regards, > > Henk Hesselink > > > On 08-02-11 02:33, Chris Stone wrote: >> >> Ovidiu, >> >> On Mon, Feb 7, 2011 at 4:19 PM, Ovidiu Sas<o...@voipembedded.com> wrote: >>> >>> By default, opensips does not modify the SDP. >>> Double check your config. If you don't need to touch SDP, make sure >>> that you are not loading nathelper or mediaproxy. Those are the two >>> modules that are changing SDP. >> >> Made sure neither of these were being loaded and used - mediaproxy >> was, but nathelper was not. I need neither, so removed, restarted >> opensips, tested a call. No change - problem persisted. So, dropped >> down to a bare config: >> >> #----------------------------------------------------------------------- >> debug=9 # debug level (cmd line: -dddddddddd) >> fork=yes >> log_stderror=no # (cmd line: -E) >> >> children=25 >> check_via=no # (cmd. line: -v) >> dns=off # (cmd. line: -r) >> rev_dns=off # (cmd. line: -R) >> port=5060 >> >> # for more info: sip_router -h >> >> # ------------------ module loading ---------------------------------- >> mpath="/usr/lib64/opensips/modules" >> >> # ----------------- setting module-specific parameters --------------- >> >> >> route{ >> forward("67.212.153.179"); >> exit; >> } >> #----------------------------------------------------------------------- >> >> Restarted OpenSIPS with the above, and problem persists - SDP routing >> modified (apparently) and Opensips proxies the audio. >> >> Incoming from upstream: >> >> INVITE sip:17204497101@67.212.153.178:5060;transport=udp SIP/2.0\r\n >> From: "STONE C AND C" >> >> <sip:+13038382386@208.94.157.10:5060>;tag=a9d5ed0-13c4-4d509b92-1bc5e644-648c7598\r\n >> To:<sip:17204497101@67.212.153.178:5060>\r\n >> Call-ID: >> CXC-260-758763d0-a9d5ed0-13c4-4d509b92-1bc5e643-43d2bf0a@208.94.157.10\r\n >> CSeq: 1 INVITE\r\n >> Via: SIP/2.0/UDP >> 208.94.157.10:5060;branch=z9hG4bK-11150e-4d509b92-1bc5e644-273291f1\r\n >> Max-Forwards: 69\r\n >> P-Asserted-Identity: "STONE C AND C " >> <sip:+13038382...@cxc.dashcs.com:5060>\r\n >> Supported: timer,100rel\r\n >> Content-Disposition: session;handling=required\r\n >> >> Contact:<sip:+13038382386@208.94.157.10:5060;maddr=208.94.157.10;transport=udp>\r\n >> Session-Expires: 1800\r\n >> Content-Type: application/sdp\r\n >> Content-Length: 238\r\n >> \r\n >> v=0\r\n >> o=Acme_UAS 0 1 IN IP4 208.94.157.10\r\n >> s=SIP Media Capabilities\r\n >> c=IN IP4 208.94.157.10\r\n >> t=0 0\r\n >> m=audio 22684 RTP/AVP 0 18 101\r\n >> a=rtpmap:0 PCMU/8000\r\n >> a=rtpmap:18 G729/8000\r\n >> a=rtpmap:101 telephone-event/8000\r\n >> a=maxptime:20\r\n >> a=sendrecv\r\n >> >> Outgoing to Asterisk: >> >> INVITE sip:17204497101@67.212.153.178:5060;transport=udp SIP/2.0\r\n >> From: "STONE C AND C" >> >> <sip:+13038382386@208.94.157.10:5060>;tag=a9d5ed0-13c4-4d509b92-1bc5e644-648c7598\r\n >> To:<sip:17204497101@67.212.153.178:5060>\r\n >> Call-ID: >> CXC-260-758763d0-a9d5ed0-13c4-4d509b92-1bc5e643-43d2bf0a@208.94.157.10\r\n >> CSeq: 1 INVITE\r\n >> Via: SIP/2.0/UDP >> 67.212.153.178:5060;branch=z9hG4bK-11150e-4d509b92-1bc5e644-273291f1\r\n >> Via: SIP/2.0/UDP >> 208.94.157.10:5060;branch=z9hG4bK-11150e-4d509b92-1bc5e644-273291f1\r\n >> Max-Forwards: 69\r\n >> P-Asserted-Identity: "STONE C AND C " >> <sip:+13038382...@cxc.dashcs.com:5060>\r\n >> Supported: timer,100rel\r\n >> Content-Disposition: session;handling=required\r\n >> >> Contact:<sip:+13038382386@67.212.153.178:5060;maddr=208.94.157.10;transport=udp>\r\n >> Session-Expires: 1800\r\n >> Content-Type: application/sdp\r\n >> Content-Length: 240\r\n >> \r\n >> v=0\r\n >> o=Acme_UAS 0 1 IN IP4 67.212.153.178\r\n >> s=SIP Media Capabilities\r\n >> c=IN IP4 67.212.153.178\r\n >> t=0 0\r\n >> m=audio 22684 RTP/AVP 0 18 101\r\n >> a=rtpmap:0 PCMU/8000\r\n >> a=rtpmap:18 G729/8000\r\n >> a=rtpmap:101 telephone-event/8000\r\n >> a=maxptime:20\r\n >> a=sendrecv\r\n >> >> I've got to be missing something stupid - the setup works great under >> 1.4 - would expect as well or better under 1.6 - but appears that >> there's some config option or default that I'm missing.... >> >> But, with such a basic config as above, not sure what it would >> be.....Would sure seem that, by some default, OpenSIPS proxies the >> audio, no? >> >> >> Thanks! >> >> >> Chris >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >
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