Anthony,<div><br></div><div>Apologies for the delay in responding here - i&#39;ve only just got around to testing this more. After finding a 404 response to BYE, I appear to have managed to fix this by changing the port the Line1 component of the ATA runs on to something other than 5060. Looks like everything is working now.</div>
<div><br></div><div>
Regards,<br>Ben Langfeld<br><br>Wave &gt; Email<br>
<br><br><div class="gmail_quote">On Fri, Feb 26, 2010 at 8:01 PM, Anthony Minessale <span dir="ltr">&lt;<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

you would have to be more specific.<br>We would need to have you test this on latest SVN trunk with a full debug log.<br><br>enter the following at your cli and reproduce the call saving all the output.<br><br>console loglevel debug<br>


sofia profile internal siptrace on<br><br><br><br><div class="gmail_quote"><div><div></div><div>On Fri, Feb 26, 2010 at 12:41 PM, Ben Langfeld <span dir="ltr">&lt;<a href="mailto:ben@langfeld.co.uk" target="_blank">ben@langfeld.co.uk</a>&gt;</span> wrote:<br>


</div></div><blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex"><div><div></div><div><span style="font-family:arial,sans-serif;font-size:13px;border-collapse:collapse">Hey,<div>


<br></div><div>I have a small 7 seat mostly softphone based PBX installation in place using freeswitch and a couple of SPA3000s for PSTN termination. Currently, aside from PSTN HUP detection, outbound calls are fine setup to directly dial the SPAs. When the internal VoIP side hangs up, freeswitch sends NORMAL_CLEARING to the ATA and the ATA drops the call. Lovely.</div>



<div><br></div><div>Unfortunately inbound calls aren&#39;t so successful. Calls are sent using the SPA dialplan to an internal freeswitch extension, and the calls connect fine. This time, when the VoIP side hangs up, no NORMAL_CLEARING is sent to the ATA. Is there a reason for this?</div>



<div><br></div><div>Can anyone give me any idea how I can get freeswitch to instruct the SPA to drop the line?</div><div><br clear="all">Regards,<br>Ben Langfeld<br><br>Wave &gt; Email</div></span>
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