-MC

On Thu, Jun 16, 2011 at 1:27 PM, Dave <dave@clancysystems.com> wrote:

Yes, I am only testing inbound. When I call from our office phone (not connected to FreeSWITCH, it's on a trunk from broadvox)  I Get a busy signal and at that moment FreeSWITCH shows...

"SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP.

If I call from my cell phone I get the Operator Message "your call did not go through" and at that moment the same thing shows in FreeSWITCH

"SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP.

----- Original Message -----

From: Michael Collins

To: FreeSWITCH Users Help

Sent: Thursday, June 16, 2011 2:11 PM

Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication

On Thu, Jun 16, 2011 at 1:02 PM, Dave <dave@clancysystems.com> wrote:

Under "domains" in acl.conf.xml I added

   <node type="allow" value="xxx.xxx.xxx.xxx"/>

The correct syntax is:

<node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>

x's are the IP address it's just a single address not a CIDR.

And I created the file velocity_did.xml with the following code..

<include>
<extension name="velocity_did">

    <condition field="destination_number" expression="^(2064001950)$">
      <action application="set" data="domain_name=$${domain}"/>
      <action application="answer"/>

   <action application="lua" data="cme_ivr.lua channel_name"/>

</condition>
</extension>
</include>

I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more.

Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that.

For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them.

-MC

Do I need to change the line ..

<action application="set" data="domain_name=$${domain}"/>

to the IP address? If so what syntax?

They have no Domain Name.

Thanks again for your help.

Dave Goodwin

----- Original Message -----

From: Michael Collins

To: FreeSWITCH Users Help

Sent: Thursday, June 16, 2011 12:15 PM

Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication

Dave,

If you are simply handling incoming calls then you will need to do two things:

Add Velocity's IP addr to the domains section of acl.conf.xml

Add an extension to the public context (conf/dialplan/public.xml)

Allowing a call into FS via an ACL will send it into the "public" context; from there you need to transfer it to the default context or just send it straight to your Lua script.

Also, I don't believe you need to use a "bridge" app based on the description you gave. Bridge is used to create a new outbound call leg (B leg) and connect it to the inbound call leg (A leg). If you just are handling a call with an IVR then there is no B leg needed, thus no need to bridge.

Welcome to FreeSWITCH, btw! :) I can highly recommend the FS book (the "bridge" book, ironically) as it discusses some of these basic concepts that will make your FS experience a whole lot more pleasant.

-MC

On Thu, Jun 16, 2011 at 8:54 AM, Dave <dave@clancysystems.com> wrote:

Hi,

I am new to FreeSWITCH and need to set up FreeSWITCH to receive calls through a Velocity Networks SIP trunk. They say I should not authenticate with Username and Password. Rather, I connect directly to the their IP address.

I put

<extension name="velocity">
      <condition field="destination_number" expression="^2064001950$">
    <action application="bridge" data="sofia/external/$ {destination_number}@their_ip_address"/>
        <action application="lua" data="cme_ivr.lua channel_name"/>
     </condition>
</extension>

in the default.xml under Dialplan

When I Dial the DID from a Phone (not one connected to the FreeSWITCH server) I get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP] from their IP, but nothing more.

What I hope to do is take the incoming call from 2064001950 and route it to the Lua IVR script above.

I Appreciate any help you may offer.

Dave Goodwin

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