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

Fri Jun 17 01:22:50 MSD 2011

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 <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *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 <msc at freeswitch.org>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>   *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 <msc at freeswitch.org>
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *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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>>>>
>>>
>>> _______________________________________________
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>>
>> _______________________________________________
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>>
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>>
>
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>
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