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Sorry to dredge up an old topic, but could someone help me with this?<br>

<br>

I need to accept and forward a call from a range of ip addresses

without any other authentication. (from-internal)<br>

<br>

Does anyone have a small snipped of extensions.conf and sip.conf that I

can use to implement this?<br>

<br>

Thanks in Advance<br>

Aaron<br>

<br>

Alexander Lopez wrote:

<blockquote

 cite="midE918F2FD95450648B7F8C957D92D527142263A@exmail.corp.opsys.com"

 type="cite">

  <pre wrap="">Exten =&gt; 123,1,NoOp(${SIPCHANINFO(recvip)})

  </pre>

  <blockquote type="cite">

    <pre wrap="">-----Original Message-----

From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> 

[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Sam Tam

Sent: Friday, February 10, 2006 3:38 AM

To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Subject: RE: [Asterisk-Users] IP Authorization

Ah that is from the CLI but still unclear about how to setup 

the extension.conf or etc..

Sam

-----Original Message-----

From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>

[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of 

Alexander Lopez

Sent: Friday, February 10, 2006 1:56 PM

To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: RE: [Asterisk-Users] IP Authorization

You can use the following:

switch3*CLI&gt; show function SIPCHANINFO

switch3*CLI&gt;

  -= Info about function 'SIPCHANINFO' =-

[Syntax]

SIPCHANINFO(item)

[Synopsis]

Gets the specified SIP parameter from the current channel

[Description]

Valid items are:

- peerip                The IP address of the peer.

- recvip                The source IP address of the peer.

- from                  The URI from the From: header.

- uri                   The URI from the Contact: header.

- useragent             The useragent.

- peername              The name of the peer.

All the info you need is there.....

    </pre>

    <blockquote type="cite">

      <pre wrap="">-----Original Message-----

From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>

[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf 

      </pre>

    </blockquote>

    <pre wrap="">Of Sam Tam

    </pre>

    <blockquote type="cite">

      <pre wrap="">Sent: Thursday, February 09, 2006 9:16 PM

To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Subject: RE: [Asterisk-Users] IP Authorization

Can you be more detail about the setup?

Sam

-----Original Message-----

From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>

[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf 

      </pre>

    </blockquote>

    <pre wrap="">Of Olle E 

    </pre>

    <blockquote type="cite">

      <pre wrap="">Johansson

Sent: Friday, February 10, 2006 4:26 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [Asterisk-Users] IP Authorization

Sam Tam wrote:

      </pre>

      <blockquote type="cite">

        <pre wrap="">I think this is a question that has been discussed before.

But you see nowadays most carriers will provide thing like

        </pre>

      </blockquote>

      <pre wrap="">SIP using

      </pre>

      <blockquote type="cite">

        <pre wrap="">IP authorization rather than username and password and I am now 

wondering whether Asterisk can do something like that or not?

        </pre>

      </blockquote>

      <pre wrap="">In the voip channels as well as in manager you can set ACLs for the 

connections you define.

/O

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