Hi,
I have installed pbxnsip on windows 2000 and enabled IPv6 support in the system. I was able to register two linphone softphones which support IPv6 to this pbxnsip server. But when I make a call I see this information.
1) Via and contact port encoded by pbxnsip while forwarding the INVITE request to callee has IP information as 0.0.0.0:5060
2) The connecction information in SDP is also 0.0.0.0
But then I added Ports|Bind to specific IP address (IPv6)| the IPv6 address. Then connection information in SDP wass sent properly. Still the Via and Contact information is incorrect. Please let me know what to do. Here is the log file.
>>>>>>>>Incoming Invite>>>>>>>>>>>>>>>>>
INVITE sip:41@[fe80::240:5ff:fe74:f12e]:5060 SIP/2.0
Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK29138
From: <sip:43@[fe80::240:5ff:fe74:f12e]>;tag=24992
To: <sip:41@[fe80::240:5ff:fe74:f12e]:5060>
Call-ID: 24976
CSeq: 21 INVITE
Contact: <sip:43@[fe80::21c:23ff:fe31:d972]:5060>
Authorization: Digest username="43", realm="[fe80::240:5ff:fe74:f12e]", nonce="52cf4dce64b277b3f1ef4a5b30cdf84a", uri="sip:41@[fe80::240:5ff:fe74:f12e]:5060", response="8aa3e734473d9bac6fdebf98553b6ccf", algorithm=MD5
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: Linphone/3.2.1 (eXosip2/3.3.0)
Subject: Phone call
Content-Length: 291
v=0
o=43 123456 654321 IN IP6 ::
s=A conversation
c=IN IP6 ::
t=0 0
m=audio 7078 RTP/AVP 8 111 110 0 3 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:111 speex/16000/1
a=rtpmap:110 speex/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
[8] 2010/04/07 00:38:34: Tagging request with existing tag
[6] 2010/04/07 00:38:34: Sending RTP for 24976#749e7695e1 to [::]:7078
[9] 2010/04/07 00:38:34: Resolve 79: udp fe80::21c:23ff:fe31:d972 5060
[7] 2010/04/07 00:38:34: SIP Tx udp:[fe80::21c:23ff:fe31:d972]:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP [::1]:5060;branch=z9hG4bK29138;rport=5060;received=fe80::21c:23ff:fe31:d972
From: <sip:43@[fe80::240:5ff:fe74:f12e]>;tag=24992
To: <sip:41@[fe80::240:5ff:fe74:f12e]:5060>;tag=749e7695e1
Call-ID: 24976
CSeq: 21 INVITE
Content-Length: 0
[7] 2010/04/07 00:38:34: Attendant: Calling extension 41
[8] 2010/04/07 00:38:34: Play audio_moh/noise.wav
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:54602
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:54603
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:53408
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:53409
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:49478
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:49479
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:50964
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:50965
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:50004
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:50005
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:60090
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:60091
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:62604
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:62605
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:50620
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:50621
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:57896
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:57897
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:59684
[9] 2010/04/07 00:38:34: UDP: Opening socket on 0.0.0.0:59685
[1] 2010/04/07 00:38:34: Could not allocate new ports!
[9] 2010/04/07 00:38:34: Using outbound proxy sip:[fe80::21c:23ff:fe31:da5a]:5060;transport=udp because of flow-label
[9] 2010/04/07 00:38:34: Resolve 80: url sip:[fe80::21c:23ff:fe31:da5a]:5060;transport=udp
[9] 2010/04/07 00:38:34: Resolve 80: a udp [fe80::21c:23ff:fe31:da5a] 5060
[9] 2010/04/07 00:38:34: Resolve 80: udp [fe80::21c:23ff:fe31:da5a] 5060
[7] 2010/04/07 00:38:34: SIP Tx udp:[fe80::21c:23ff:fe31:da5a]:5060:
INVITE sip:41@[fe80::21c:23ff:fe31:da5a]:5060;line=254fef6d183784e SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK-bffbb9ccef010983980a78d9dcf7aa76;rport
From: "Fourty Three" <sip:43@pbx.company.com>;tag=23300
To: "Fourty One" <sip:41@pbx.company.com>
Call-ID: 3c35ce52@pbx
CSeq: 840 INVITE
Max-Forwards: 70
Contact: <sip:41@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/3.4.0.3201
Alert-Info: <http://127.0.0.1/Bellcore-dr2>
Content-Type: application/sdp
Content-Length: 359
v=0
o=- 29657 29657 IN IP6 fe80::240:5ff:fe74:f12e
s=-
c=IN IP6 fe80::240:5ff:fe74:f12e
t=0 0
m=audio 59684 RTP/AVP 0 8 9 18 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
Look at the outgoing Invite (the above one)
