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

David Ponzone david.ponzone at ipeva.fr
Fri Jun 17 09:57:35 MSD 2011


Dave,

if you use the default config, I would recommend you configure Vitelity to send you the packets to your port 5080 (profile external).
Auth is disabled on external, so I think this port is more adapted to be used for unauth trunks.

David Ponzone  Direction Technique
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Le 16/06/2011 à 23:46, Dave a écrit :

> Will Do. Thanks very much for your help.  I bought the book:)
>  
> Dave
>  
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Thursday, June 16, 2011 3:22 PM
> Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
> 
> Okay, I want you to go read up on gather data and dropping it on pastebin. The skills you hone there will server you well. :)
> 
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
> 
> Need a debug log and a sip trace. Drop it on pastebin. Hint: use "sofia global siptrace on" to get all sip traffic logged to the console.
> 
> Put the pb link in this email thread.
> -MC
> 
> On Thu, Jun 16, 2011 at 2:18 PM, Dave <dave at clancysystems.com> wrote:
> I did both, and even restarted the service. same results. Velocity did a call capture on thier end and sent it to me. They say that FreeSWITCH is requiring proxy authentication and that that is the issue.
> SIP Status: 407 Proxy Authentication Required
>  
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Thursday, June 16, 2011 2:46 PM
> Subject: Re: [Freeswitch-users] Setting Up an Sip Trunk without Authentication
> 
> Did you reloadxml and restart the sofia profile after the changes? Also, you need to "reloadacl" after doing a change to acl.conf.xml.
> 
> -MC
> 
> On Thu, Jun 16, 2011 at 1:27 PM, Dave <dave at 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 at 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 at 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 at 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=$${domain}"/>
>       <action application="answer"/>
> 
>    <action application="lua" data="cme_ivr.lua channel_name"/>
>  
>  </condition>
>   </extension>
> </include>
> I still just get the "SIP auth challenge (INVITE) on sofia profile 'internal' for [myphone at theirIP] from their IP, but nothing more.
>  
> Wait - can you explain *exactly* what you're doing for testing? Are you trying to dial out via Velocity and back in to your DID? If so then you have 2 completely different things you need to set up. The instructions I gave were only for inbound DID, so use a cell phone or something to test that.
> 
> For outbound it seems like they are sending you an auth challenge, which means they need to give you a username and password as well as a host/ip. You need to create a gateway, preferably in your external profile. Just be sure to set the "register" param to false since they are not expecting you to register with them. 
> 
> -MC
> 
> Do I need to change the line ..
>  
> <action application="set" data="domain_name=$${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 at 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 at 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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