Hello Brian and Everyone,<div><br></div><div><br></div><div>Here is a link to the siptrace output. Sorry for the delay, it was my birthday and I was away for the weekend.</div><div><br></div><div><br></div><div><a href="http://pastebin.freeswitch.org/19057">http://pastebin.freeswitch.org/19057</a>
</div><div><br></div><div><br></div><div>Thanks,</div><div><br></div><div>-Chris</div><div><br></div><div><br></div><div>
<br><br><div class="gmail_quote">On Thu, May 10, 2012 at 10:16 PM, Brian Foster <span dir="ltr">&lt;<a href="mailto:bdfoster@endigotech.com" target="_blank">bdfoster@endigotech.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<p>1. Run fs_cli and do &#39;sofia global siptrace on&#39; without quotes. Stay in the cli and run a call in an instance where it would fail. Make sure your scroll buffer on your SSH terminal (PuTTY or what ever you have) to unlimited or something bog like 20K lines. Copy the logs out and put them on <a href="http://pastebin.freeswitch.org" target="_blank">http://pastebin.freeswitch.org</a>. If prompted for a username/password, please read the dialog box. SpanDSP doesn&#39;t have a great ability to be degugged on the click but it will give is something to go on.</p>



<p>2. There usually isn&#39;t much you can do as far as ensuring the VPS is capable of keeping the timing right. VoIP is reliant heavily on a good timing source. Faxing is even more dependent.</p>
<p>I&#39;m not really sure what the MOS score is useful for, maybe someone else can chime in on that.</p>
<p>Something you need to make sure you are doing (specifically with flowroute) is making sure you are doing a t.38 reinvite. Flowroute does not auto detect t.38. Also freeswitch might have to transcode from audio ulaw/alaw to t.38 if your FAX machine doesn&#39;t actually support t.38. </p>



<p>I don&#39;t really know much about the new SPA 112/122&#39;s but usually the ATA has a t.38 passthru mode that is auto detected. That might be what you are referring to.</p>
<p>-BDF</p><div class="HOEnZb"><div class="h5">
<div class="gmail_quote">On May 10, 2012 2:35 PM, &quot;Chris Ferreira&quot; &lt;<a href="mailto:chris@gonumina.com" target="_blank">chris@gonumina.com</a>&gt; wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">


Hi Bryan,<div><br></div><div><br></div><div>Thanks for your response, much appreciated.</div><div><br></div><div><br></div><div>1. I have never had to do a debug other than analyze fs_cli output and iv&#39;e never done a sip trace. I have started reading up on how to do these on the Wiki and will generate some files to take a look at.</div>




<div><br></div><div><br></div><div>2. I have heard this before about VPS&#39;s and even VMWare servers running on local physical hardware. I hope to be able to overcome these issues.</div><div><br></div><div><br></div><div>




3. Other calls work fine throughout the switch and between carriers.</div><div><br></div><div><br></div><div><br></div><div>I don&#39;t know if this is a good actual measure of anything, but VoIPSpear.com says that the servers MOS Score is a 4.0 .</div>




<div><br></div><div><br></div><div>Other than the server running on a VPS, is my setup with the ATA registering to FreeSWITCH as an intermediary between Flowroute.com a correct practice? Or is there something else I could be doing?</div>




<div><br></div><div><br></div><div>I know that if I was not needing to use an Analog Fax machine for sending/receiving that I would have other options. But unfortunately the Dinosaur must have it&#39;s Dial Tone.</div><div>




<br></div><div><br></div><div><br></div><div>Thanks,</div><div><br></div><div>-Chris<br><font size="3"><br></font><br>
<br><br><div class="gmail_quote">On Thu, May 10, 2012 at 2:03 PM, Brian Foster <span dir="ltr">&lt;<a href="mailto:bdfoster@endigotech.com" target="_blank">bdfoster@endigotech.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">




1. Debug tells all. We need the siptrace as well. <div><br></div><div>2. VPS&#39;s are usually bad at sending/receiving faxes even if the media is just being routed through it. </div><div><br></div><div>3. You could also look at session timers. Are other calls working?</div>





<div><br></div><div>The only real way to know is if we have number 1 in our possession, otherwise we just start guessing.</div><div><br></div><div>-BDF<br><br><div class="gmail_quote"><div><div>On Thu, May 10, 2012 at 1:30 PM, Chris Ferreira <span dir="ltr">&lt;<a href="mailto:chris@gonumina.com" target="_blank">chris@gonumina.com</a>&gt;</span> wrote:<br>





</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>Hello All,<div><br></div><div><br></div><div><br></div><div>I have a FreeSWITCH install running on CentOS 5.6 on a Linode VPS. I am trying to get my SPA112 (Version 1.1 Firmware) to send faxes (and eventually receive) successfully. I have the ATA registered as an extension and it&#39;s primary outgoing route is through Flowroute. I have T.38 enabled on the ATA and I have diabled ECM on the analog fax machine. All of the CDR&#39;s in Flowroute show that the test faxes are all ending their calls at 39 seconds.
</div><div><br></div><div><br></div><div>I can post screen shots of the ATA config or provide any other config info.</div><div><br></div><div><br></div><div>Is there something I am missing, or should this setup work? </div>







<div><br></div><div>I have poked around for a while for info, but unlike the PAP2T there is little info on these SPA112&#39;s. This is my first post asking for help as I usually try to resolve things on my own. But for this, I will defer to everyone&#39;s experience.</div>








<div><br></div><div><br></div><div><br></div><div>Thanks,</div><div><br></div><div>-Chris</div>
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