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Jerry777

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  1. Here is the output from the "route -A inet6" command: Kernel IPv6 routing table Destination Next Hop Flags Metric Ref Use Iface ::1/128 * U 0 10 2 lo fe80::204:23ff:fea7:d60a/128 * U 0 34856 2 lo fe80::204:8dff:feff:fffe/128 fe80::204:8dff:feff:fffe UC 0 1 1 eth1 fe80::/64 * U 256 0 0 eth1 ff00::/8 * U 256 0 0 eth1 Best Regards
  2. Correction: Paragraph two of my last reply should have said "IPv6" instead of "IPv4", as follows: However, in the SIP header portion of the INVITE I am still seeing 0.0.0.0 in the Contact and Via SIP headers. So I have not found a way to cause this to be populated with the valid IPv6 address. Thanks
  3. We are using Red Hat 3.4.6-3. I did find a way to get it to set the SDP address correctly, via the SystemAdministration|Ports|Rtp:*|BindtospecificIPaddress(IPv6) menu. I set this field to the workstation's IPv6 IP address and now that address is appearing in the SDP. However, in the SIP header portion of the INVITE I am still seeing 0.0.0.0 in the Contact and Via SIP headers. So I have not found a way to cause this to be populated with the valid IPv4 address. I presume the routing table is correct, because messages are sent/received correctly from/to the workstation's IPv6 address. It's only the SIP header content that has the issue. Best Regards
  4. Also, notice the pbxnsip server's contact address is wrong (0.0.0.0). It should be it's IPv6 address. Best Regards
  5. Here is the pbxnsip log when I make a call from line 44 to 45 (both registered with IPv6 addresses). Notice that the pbxnsip server is sending SDP to the far-end with an IP6 address of 0.0.0.0. I can provide follow-on messages, if needed. Any idea why this is happening? Logfile Clear or Reload the log. [9] 2009/06/30 09:08:48: Remote site closed the connection [9] 2009/06/30 09:08:56: SIP Rx udp:[fe80::204:8dff:feff:fffe]:5060: INVITE sip:45@[fe80::204:23ff:fea7:d60a]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP [fe80::204:8dff:feff:fffe];rport;branch=z9hG4bKN24Qa3ert8NSB Max-Forwards: 70 From: Jerry's 44 <sip:44@[fe80::204:23ff:fea7:d60a]:5060;user=phone>;tag=jm9pcvga1BDyD To: <sip:45@[fe80::204:23ff:fea7:d60a]:5060>;user=phone Call-ID: 902f30db-dfe8-122c-a4a6-09ac0212a3be CSeq: 117054713 INVITE Contact: Jerry's 44 <sip:44@[fe80::204:8dff:feff:fffe]:5060;transport=udp> User-Agent: TONE COMMANDER 7810/fffffe Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: replaces, callerid, 100rel, resource-priority, sdp-anat Allow-Events: talk, hold, refer, call-info, dialog, sla, include-session-description Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 266 P-Early-Media: supported P-Preferred-Identity: <sip:44@[fe80::204:23ff:fea7:d60a]> v=0 o=TC 722710983 722710983 IN IP6 fe80::204:8dff:feff:fffe s=session c=IN IP6 fe80::204:8dff:feff:fffe t=0 0 m=audio 1760 RTP/AVP 0 18 4 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000/1 [9] 2009/06/30 09:08:56: UDP: Opening socket on port 52870 [9] 2009/06/30 09:08:56: UDP: Opening socket on port 52871 [9] 2009/06/30 09:08:56: UDPv6: Opening socket on port 52870 [9] 2009/06/30 09:08:56: UDPv6: Opening socket on port 52871 [8] 2009/06/30 09:08:56: Could not find a trunk (1 trunks) [9] 2009/06/30 09:08:56: Resolve 20871: aaaa udp fe80::204:8dff:feff:fffe 5060 [9] 2009/06/30 09:08:56: Resolve 20871: udp fe80::204:8dff:feff:fffe 5060 [9] 2009/06/30 09:08:56: SIP Tx udp:[fe80::204:8dff:feff:fffe]:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP [fe80::204:8dff:feff:fffe];rport=5060;branch=z9hG4bKN24Qa3ert8NSB From: Jerry's 44 <sip:44@[fe80::204:23ff:fea7:d60a]:5060;user=phone>;tag=jm9pcvga1BDyD To: <sip:45@[fe80::204:23ff:fea7:d60a]:5060>;tag=1a053bc5c1;user=phone Call-ID: 902f30db-dfe8-122c-a4a6-09ac0212a3be CSeq: 117054713 INVITE Content-Length: 0 [7] 2009/06/30 09:08:56: Set packet length to 20 [7] 2009/06/30 09:08:56: Last message repeated 2 times [6] 2009/06/30 09:08:56: Sending RTP for 902f30db-dfe8-122c-a4a6-09ac0212a3be#1a053bc5c1 to [fe80::204:8dff:feff:fffe]:1760 [7] 2009/06/30 09:08:56: Attendant: Calling extension 45 [7] 2009/06/30 09:08:56: Cannot convert number 44 into global format [8] 2009/06/30 09:08:56: Play audio_moh/noise.wav [9] 2009/06/30 09:08:56: UDP: Opening socket on port 61172 [9] 2009/06/30 09:08:56: UDP: Opening socket on port 61173 [9] 2009/06/30 09:08:56: UDPv6: Opening socket on port 61172 [9] 2009/06/30 09:08:56: UDPv6: Opening socket on port 61173 [9] 2009/06/30 09:08:56: Using outbound proxy sip:[fe80::204:8dff:feff:ffe4]:5060;transport=udp because of flow-label [9] 2009/06/30 09:08:56: Resolve 20872: url sip:[fe80::204:8dff:feff:ffe4]:5060;transport=udp [9] 2009/06/30 09:08:56: Resolve 20872: a udp [fe80::204:8dff:feff:ffe4] 5060 [9] 2009/06/30 09:08:56: Resolve 20872: udp [fe80::204:8dff:feff:ffe4] 5060 [9] 2009/06/30 09:08:56: SIP Tx udp:[fe80::204:8dff:feff:ffe4]:5060: INVITE sip:45@[fe80::204:8dff:feff:ffe4]:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK-e1523731b2cb1200ab2fa761491d7109;rport From: "Fourty Four" <sip:44@pbx.company.com>;tag=1142770007 To: "Fourty Five" <sip:45@pbx.company.com> Call-ID: c3852a9b@pbx CSeq: 9340 INVITE Max-Forwards: 70 Contact: <sip:45@0.0.0.0:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.1.3023 Alert-Info: <http://127.0.0.1/Bellcore-dr2> Content-Type: application/sdp Content-Length: 286 v=0 o=- 75511103 75511103 IN IP6 0.0.0.0 s=- c=IN IP6 0.0.0.0 t=0 0 m=audio 61172 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv
  6. We installed pbxnsip on a Red Hat 3.4.6-3 workstation.
  7. Does pbxnsip support IPv6 SDP? I have version 3 of pbxnsip. It allows IPv6 addressing for SIP signaling, but does not appear to support IPv6 addresses inside the SDP body, so calls between two IPv6 VoIP phones have no talk path. I noticed that when forwarding INVITEs from the caller to the callee, pbxnsip indicates the connection address is IP6, however, the address indicated is 0.0.0.0 which is a null IP4 address. Could this be my pbxnsip configuration? Or is it not supported yet? If it will be supported, is there an estimate of when this will be available? Best Regards
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