Some additional debug output:
<--- SIP read from 10.0.2.136:5060 --->
INVITE sip:103 at 10.0.4.147 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.136:5060;branch=z9hG4bKb690d74021ef56
From: sip:paul at 10.0.4.147;tag=C1431100-8D08-67CF-A5B0-EDB815EEBF60
To: <sip:103 at 10.0.4.147>
Max-Forwards: 70
CSeq: 1 INVITE
Call-ID: C1431100-8D08-8293-CAB6-5A32B3B58C6C-5060 at 10.0.2.136
Contact: <sip:vgp at 10.0.2.136:5060>
Content-Length: 218
Content-Type: application/sdp
Supported: timer
v=0
o=VoiceGenie 1073962426 1 IN IP4 10.0.2.136
s=phone-call
c=IN IP4 10.0.2.136
t=0 0
m=audio 1032 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (11 headers 10 lines) ---
Sending to 10.0.2.136 : 5060 (no NAT)
Using INVITE request as basis request -
C1431100-8D08-8293-CAB6-5A32B3B58C6C-5060 at 10.0.2.136
<--- Reliably Transmitting (no NAT) to 10.0.2.136:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.0.2.136:5060;branch=z9hG4bKb690d74021ef56;received=10.0.2.136
From: sip:paul at 10.0.4.147;tag=C1431100-8D08-67CF-A5B0-EDB815EEBF60
To: <sip:103 at 10.0.4.147>;tag=as64474d5c
Call-ID: C1431100-8D08-8293-CAB6-5A32B3B58C6C-5060 at 10.0.2.136
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="277024dd"
Content-Length: 0
________________________________
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paul
Campbell
Sent: 26 October 2007 13:51
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Still more auth problems
Firstly can I ask when the documentation site will be online again? I'm
struggling here without it.
Further to my recent post I have tried to simplify things a little.
I have used a VoiceXML app to simple call an asterisk extension. EG:
<form id="transfer">
<block>
<call name="xfer" dest="sip:101 at
10.0.4.147:5060"/>
<if cond="xfer == 'connected'">
<prompt>Call connected</prompt>
<elseif cond="xfer == 'noanswer'"/>
<prompt>There is no answer</prompt>
<else/>
<prompt>Call is not connected.
Return value is <value expr="xfer"/>
</prompt>
</if>
</block>
</form>
If I just put "sip:101" as dest then I get silence and "There is
no
answer", but the asterisk extension does not ring. If I put the dest as
above I just get "Call is not connected, Return value is Failed."
Asterisk insists on saying "407 Proxy Authentication Required"
I don't understand. I'm not asking it to proxy anything. I'm
simply
putting a call through to an extension.
The Voice server is registered with Asterisk as a friend. I've tried
various options like:
Allowanonymousproxy=yes
Allowanonymoussipcalls=yes
And
Insecure=very
Yet I can ONLY get 407 Proxy Authentication required.
Can anyone give me even a hint in the right direction?
Thanks
Paul
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