I think it can (inbound). Outbound with sipxbridge would in turn introduce asynchronous routing but inbound via IP will work.
-- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! On Sep 27, 2012 6:58 PM, "Michael Picher" <[email protected]> wrote: > 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. > > > _______________________________________________ > sipx-dev mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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