[asterisk-users] Still more auth problems

[Paul Campbell]

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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