how can the 1 server have 2 different outside IP addresses? i don't think sipxbridge / media relay can deal with this.
i'd setup another box with sipxbridge on it also and the different outside IP. Then do 1-to-1 nat on the firewall from that outside IP to this other box. mike On Thu, Sep 27, 2012 at 6:42 PM, Gerald Drouillard <[email protected]>wrote: > Still looks like there may be authentication issues. We have an > installation that has a firewall that has 2 different ISP's and public > static IP addresses. We are seeing random inbound calls get rejected on > the primary public IP of the server. The calls are coming from > voiceinnovations (VI) on 5080 (using IP auth). We have our backup IP > address setup with VI. VI call fails on the primary then tries on the > backup IP and the call comes in. 192.168.0.2 is the IP of the sipxecs > server. Most the other IP have been changed along with the phone numbers. > > We have sent the profiles to the server. Sent profiles to the gateway. > Rebooted. > Here is the Invite and 500 response on the Primary IP: > > INVITE sip:[email protected]:5080 SIP/2.0 > Via: SIP/2.0/UDP 64.136.174.30:5060;branch=z9hG4bK1sansay1100463543rdb10333 > Record-Route: > <sip:[email protected]:5060;lr;transport=udp> > To: <sip:[email protected]> > From: "DROUILLARD&ASC, " > <sip:[email protected]>;tag=sansay1100463543rdb10333 > Call-ID: [email protected] > CSeq: 1 INVITE > Contact: <sip:[email protected]:5060> > Supported: timer > Session-Expires: 1800;refresher=uac > Min-SE: 90 > P-Asserted-Identity: "DROUILLARD&ASC, " <sip:[email protected]:5060> > Max-Forwards: 68 > Content-Type: application/sdp > Content-Length: 298 > > v=0 > o=Sansay-VSXi 188 1 IN IP4 64.136.174.30 > s=Session Controller > c=IN IP4 4.55.9.198 > t=0 0 > m=audio 20792 RTP/AVP 0 8 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > a=maxptime:20 > > > SIP/2.0 500 Server Internal Error > Via: SIP/2.0/UDP 64.136.174.30:5060;branch=z9hG4bK1sansay1100463543rdb10333 > To: <sip:[email protected]> > From: "DROUILLARD&ASC," > <sip:[email protected]>;tag=sansay1100463543rdb10333 > Call-ID: [email protected] > CSeq: 1 INVITE > Server: sipXecs/4.4.0 sipXecs/sipxbridge (Linux) > Content-Type: message/sipfrag > Content-Length: 103 > > Exception Info Initialization exception while processing request at > BackToBackUserAgentFactory.java:199 > > > > Here is the invite on the second IP > > INVITE sip:[email protected]:5080 SIP/2.0 > Via: SIP/2.0/UDP 64.136.174.30:5060;branch=z9hG4bK1sansay1100463570rdb11875 > Record-Route: > <sip:[email protected]:5060;lr;transport=udp> > To: <sip:[email protected]> > From: "DROUILLARD&ASC, " > <sip:[email protected]>;tag=sansay1100463570rdb11875 > Call-ID: [email protected] > CSeq: 1 INVITE > Contact: <sip:[email protected]:5060> > Supported: timer > Session-Expires: 1800;refresher=uac > Min-SE: 90 > P-Asserted-Identity: "DROUILLARD&ASC, " <sip:[email protected]:5060> > Max-Forwards: 68 > Content-Type: application/sdp > Content-Length: 298 > > v=0 > o=Sansay-VSXi 188 1 IN IP4 64.136.174.30 > s=Session Controller > c=IN IP4 4.55.9.198 > t=0 0 > m=audio 20792 RTP/AVP 0 8 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > a=maxptime:20 > > SIP/2.0 200 OK > Via: SIP/2.0/UDP 64.136.174.30:5060;branch=z9hG4bK1sansay1100463570rdb11875 > Record-Route: > <sip:[email protected]:5060;lr;transport=udp> > To: <sip:[email protected]>;tag=643886544 > From: "DROUILLARD&ASC," > <sip:[email protected]>;tag=sansay1100463570rdb11875 > Call-ID: [email protected] > CSeq: 1 INVITE > Server: sipXecs/4.4.0 sipXecs/sipxbridge (Linux) > Contact: <sip:[email protected]:5080;transport=udp> > Content-Type: application/sdp > Content-Length: 247 > > v=0 > o=sipxbridge 7129984948061286249 1 IN IP4 000.000.000.001 > s=FreeSWITCH > c=IN IP4 000.000.000.001 > t=0 0 > m=audio 30500 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > > > Here is the Invite on the Primary and It Worked > > INVITE sip:[email protected]:5080 SIP/2.0 > Via: SIP/2.0/UDP 64.136.174.30:5060;branch=z9hG4bK1sansay1100449949rdb9903 > Record-Route: > <sip:[email protected]:5060;lr;transport=udp> > To: <sip:[email protected]> > From: "DROUILLARD&ASC, " > <sip:[email protected]>;tag=sansay1100449949rdb9903 > Call-ID: [email protected] > CSeq: 1 INVITE > Contact: <sip:[email protected]:5060> > Supported: timer > Session-Expires: 1800;refresher=uac > Min-SE: 90 > P-Asserted-Identity: "DROUILLARD&ASC, " <sip:[email protected]:5060> > Max-Forwards: 66 > Content-Type: application/sdp > Content-Length: 297 > > v=0 > o=Sansay-VSXi 188 1 IN IP4 64.136.174.30 > s=Session Controller > c=IN IP4 4.55.5.66 > t=0 0 > m=audio 23066 RTP/AVP 0 8 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > a=maxptime:20 > > > -- > Regards > -------------------------------------- > Gerald Drouillard > Technology Architect > Drouillard & Associates, Inc.http://www.Drouillard.biz > > > _______________________________________________ > sipx-dev mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square**** Suite 201**** Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher <http://twitter.com/mpicher> linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro> www.ezuce.com ------------------------------------------------------------------------------------------------------------ There are 10 kinds of people in the world, those who understand binary and those who don't.
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