[Freeswitch-users] Newbie: Avaya SES <>Freeswitch 407 Proxy Authentication error

Gerry Hull gerry at pstn2.net
Tue Oct 21 07:33:25 PDT 2008


Thanks Anthony,

We have turned off all nat-related anything, but are still having an issue.
However, here is more information:

-- We are running FreeSwitch on Windows Server 2003 R2
-- We are running FreeSwitch in a console app for testing

Calls from Avaya are answered in the dial plan and a javascript is executed:


<extension name="5060">

                <condition field="destination_number" expression="^5060$">

                                <action application="javascript"
data="Answer.js 5061 welcome.wav"/>

                </condition>

</extension>

Here's the js:


var sAudioFilePath = "/sounds/";



session.answer();



if(argv.length > 1)

{

    for (var i = 1; i < argv.length; i ++)

    {

                    session.streamFile(sAudioFilePath + argv[i]);

    }

}



session.execute("transfer", argv[0]);



exit();

After playing the file, the caller is transferred into a park, using this
dialplan code:


<extension name="5060_Park">

      <condition field="destination_number" expression="^5061$">

      <action application="set"
data="fifo_music=$${sound_prefix}/sounds/pleasewait.wav"/>

                <action application="fifo" data="5061@$${domain} in"/>

<action application="set" data="hangup_after_bridge=true"/>

      </condition>

    </extension>



Here's the issue:   If we hang up the call while it is parked, we see
NOTHING in the sofia debug or freeswitch log.

(console window).   We deceided to log freeswitch log info to a file.   If
we shut down FreeSwitch when it is the state with a call that is parked, we
do see the SIP BYE message, and everythig shuts down.  It's as if some
thread is hung.



All of this only happens with an inbound Avaya trunk.   If we dial from an
anonymous SIP connection, everything works fine.



Ideas?



Regards,



Gerry

On Fri, Oct 17, 2008 at 3:30 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> make sure you have all of the nat related params off in FS just in case.
> edit your sip profile and comment out anything that says nat
>
>
>
> On Fri, Oct 17, 2008 at 12:51 PM, Gerry Hull <gerry at pstn2.net> wrote:
>
>> I've solved most of my issues by trunking Freeswitch with Avaya SES.   It
>> was more on the SES configuration side.
>> FreeSwitch seems to perform flawlessly.
>>
>> Now the only problem is, if I dial into Freeswitch from an Avaya
>> extension, and hangup the call, I get no SIP BYE or CANCEL from SES!
>>
>> Oh joy!
>>
>> Thanks for all your help on this, guys.
>>
>> Gerry
>>
>>
>> On Thu, Oct 16, 2008 at 1:48 AM, Gayatri Kulkarni <xtpl.gayatri at gmail.com
>> > wrote:
>>
>>> Gerry,
>>> Did you manage to take the ethereal trace? It would be definitely of more
>>> help and we can narrow down the actual problem
>>> Do you have access to Avaya SES?
>>>
>>> To take the ethreal trace, you should:
>>> *1)telnet* <user>@<Avaya SES>  //reduces size of the trace, ssh size
>>> usually goes in GBs
>>> 2)login
>>> 3)tethereal -i eth0 -f <path/to/filename>
>>> 4)try to make the call (get the 407)
>>> 5)kill tethereal (Ctrl+C)
>>> 6) copy the file to a place where you can sit and analyze it
>>>
>>> You can analyze the trace yourself if you have wireshark installed, or
>>> send it over
>>>
>>> --
>>> Regards,
>>> Gayatri Kulkarni
>>>
>>>
>>> On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull <gerry at pstn2.net> wrote:
>>>
>>>> Gayatri,
>>>>
>>>> Any idea on how to enable this response in Freeswitch?
>>>>
>>>> David,
>>>>
>>>> Not sure of the "lr"...
>>>>
>>>> On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni <
>>>> xtpl.gayatri at gmail.com> wrote:
>>>>
>>>>> Thanks David!
>>>>> Gerry,
>>>>> From the debug info you have sent, looks like Avaya SES asks for PAI
>>>>> i.e Proxy Authentication Indication - It's a kind of challenge response
>>>>> authentication. After it receives the user's digest in response to this
>>>>> request (again), it authenticates the user. This is the normal behavior of
>>>>> Avaya SES.
>>>>> the users' digest is not sent again it seems!
>>>>>
>>>>>   On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave at 3c.co.uk> wrote:
>>>>>
>>>>>>
>>>>>>  On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
>>>>>>
>>>>>>  Record-Route: <sip:10.0.2.154:5060;lr>
>>>>>>
>>>>>>    Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
>>>>>>
>>>>>> what's the 'lr' next to the port number?
>>>>>>
>>>>>>
>>>>>> short for 'loose routing' - see here for a bit of an explanation:
>>>>>> http://www.tech-invite.com/Ti-sip-dialog.html
>>>>>>
>>>>>> --Dave
>>>>>>
>>>>>> _______________________________________________
>>>>>> Freeswitch-users mailing list
>>>>>> Freeswitch-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Regards,
>>>>> Gayatri Kulkarni
>>>>>
>>>>> _______________________________________________
>>>>> Freeswitch-users mailing list
>>>>> Freeswitch-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
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>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
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>>>>
>>>
>>>
>>>
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>>
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>>
>
>
> --
> Anthony Minessale II
>
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