<br><br><div class="gmail_quote">Le 14 avril 2012 11:30, Ben WIlliams <span dir="ltr"><<a href="mailto:bwilliams%2Basterisk@jadeworld.com">bwilliams+asterisk@jadeworld.com</a>></span> a écrit :<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
This is a really simple problem that I just can't get to work. There<br>
are two Asterisk servers with the following sip user and peer. When a<br>
call is attempted, Asterisk</blockquote><div><br>Which instance are you talking about, here ?<br> <br></div><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
is not sending authentication details in<br>
response to the 401. Note, if the secret is blank on 172.16.0.2 test,<br>
the INVITE succeeds.<br>
<br>
on <a href="http://172.16.0.2" target="_blank">172.16.0.2</a>:<br>
<br>
[test]<br>
type=friend<br>
secret=abcde<br>
host=dynamic<br>
context=demo<br>
<br>
on 172.16.0.1 :<br>
<br>
[natty]<br>
type=peer<br>
host=172.16.0.2<br>
fromuser=test<br>
secret=abcde<br>
<br>
originate SIP/natty/1234568 extension 200<br>
== Using SIP RTP CoS mark 5<br>
Audio is at 172.16.0.1 port 19486<br>
Adding codec 0x2 (gsm) to SDP<br>
Adding codec 0x4 (ulaw) to SDP<br>
Adding codec 0x8 (alaw) to SDP<br>
Adding non-codec 0x1 (telephone-event) to SDP<br>
Reliably Transmitting (no NAT) to <a href="http://172.16.0.2:5060" target="_blank">172.16.0.2:5060</a>:<br>
INVITE <a href="mailto:sip%3A1234568@172.16.0.2">sip:1234568@172.16.0.2</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport<br>
Max-Forwards: 70<br>
From: "asterisk" <<a href="http://sip:test@172.16.0.1:5066" target="_blank">sip:test@172.16.0.1:5066</a>>;tag=as1689b2b6<br>
To: <<a href="mailto:sip%3A1234568@172.16.0.2">sip:1234568@172.16.0.2</a>><br>
Contact: <<a href="http://sip:test@172.16.0.1:5066" target="_blank">sip:test@172.16.0.1:5066</a>><br>
Call-ID: <a href="mailto:2353cf0e59596e285c684b44220f8915@172.16.0.1">2353cf0e59596e285c684b44220f8915@172.16.0.1</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1<br>
Date: Sat, 14 Apr 2012 09:10:38 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces, timer<br>
Content-Type: application/sdp<br>
Content-Length: 290<br>
<br>
v=0<br>
o=root 1594270426 1594270426 IN IP4 172.16.0.1<br>
s=Asterisk PBX 1.6.2.9-2ubuntu2.1<br>
c=IN IP4 172.16.0.1<br>
t=0 0<br>
m=audio 19486 RTP/AVP 3 0 8 101<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
---<br>
<br>
<--- SIP read from UDP:<a href="http://172.16.0.2:5060" target="_blank">172.16.0.2:5060</a> ---><br>
SIP/2.0 401 Unauthorized<br>
Via: SIP/2.0/UDP<br>
172.16.0.1:5066;branch=z9hG4bK59f50e30;received=172.16.0.1;rport=5066<br>
From: "asterisk" <<a href="http://sip:test@172.16.0.1:5066" target="_blank">sip:test@172.16.0.1:5066</a>>;tag=as1689b2b6<br>
To: <<a href="mailto:sip%3A1234568@172.16.0.2">sip:1234568@172.16.0.2</a>>;tag=as1a6c2364<br>
Call-ID: <a href="mailto:2353cf0e59596e285c684b44220f8915@172.16.0.1">2353cf0e59596e285c684b44220f8915@172.16.0.1</a><br>
CSeq: 102 INVITE<br>
Server: Asterisk PBX 1.6.2.9-2ubuntu2.1<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces, timer<br>
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a03a1d3"<br>
Content-Length: 0<br>
<br>
<br>
<-------------><br>
--- (11 headers 0 lines) ---<br>
Transmitting (no NAT) to <a href="http://172.16.0.2:5060" target="_blank">172.16.0.2:5060</a>:<br>
ACK <a href="mailto:sip%3A1234568@172.16.0.2">sip:1234568@172.16.0.2</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport<br>
Max-Forwards: 70<br>
From: "asterisk" <<a href="http://sip:test@172.16.0.1:5066" target="_blank">sip:test@172.16.0.1:5066</a>>;tag=as1689b2b6<br>
To: <<a href="mailto:sip%3A1234568@172.16.0.2">sip:1234568@172.16.0.2</a>>;tag=as1a6c2364<br>
Contact: <<a href="http://sip:test@172.16.0.1:5066" target="_blank">sip:test@172.16.0.1:5066</a>><br>
Call-ID: <a href="mailto:2353cf0e59596e285c684b44220f8915@172.16.0.1">2353cf0e59596e285c684b44220f8915@172.16.0.1</a><br>
CSeq: 102 ACK<br>
User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1<br>
Content-Length: 0<br>
<br>
<br>
---<br>
[Apr 14 21:10:38] NOTICE[31158]: chan_sip.c:17975<br>
handle_response_invite: Failed to authenticate on INVITE to<br>
'"asterisk" <<a href="http://sip:test@172.16.0.1:5066" target="_blank">sip:test@172.16.0.1:5066</a>>;tag=as1689b2b6'<br>
<br>
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</blockquote></div><br>