<div dir="ltr">I may have the answer. I think it's the Polycom phone that's doing it. Here's a SIP trace:<div><br></div><div><div>recv 780 bytes from udp/[192.168.4.252]:5060 at 01:00:16.432648:</div><div> ------------------------------------------------------------------------</div>
<div> CANCEL sip:1<my_number>@<a href="http://mydomian.com">mydomian.com</a>;user=phone SIP/2.0</div><div> Via: SIP/2.0/UDP 192.168.4.252;branch=z9hG4bKd0f2a84497538941</div><div> From: "523" <<a href="mailto:sip%3A523@mydomain.com">sip:523@mydomain.com</a>>;tag=2615BA99-7EE01046</div>
<div> To: <sip:1<my_number>@<a href="http://mydomain.com">mydomain.com</a>;user=phone></div><div> CSeq: 2 CANCEL</div><div> Call-ID: <a href="mailto:6b4e906d-5566f0da-6249477@192.168.4.252">6b4e906d-5566f0da-6249477@192.168.4.252</a></div>
<div> Contact: <<a href="mailto:sip%3A523@192.168.4.252">sip:523@192.168.4.252</a>></div><div> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER</div><div> User-Agent: PolycomSoundPointIP-SPIP_320-UA/<a href="http://3.3.5.0247">3.3.5.0247</a></div>
<div> Proxy-Authorization: Digest username="523", realm="<a href="http://mydomain.com">mydomain.com</a>", nonce="4e8ec7ff-1276-4bbc-b80c-a99f3afb4510", qop=auth, cnonce="saAiG3VwjXuGreZ", nc=00000002, uri="<a href="mailto:sip%3A14087390936@mydomian.com">sip:14087390936@mydomian.com</a>;user=phone", response="4625acb23c5776a36517032017af753c", algorithm=MD5</div>
<div> Max-Forwards: 70</div><div> Content-Length: 0</div><div><br></div><div><my_number> was substituted for the number I dialed (a POTS number). My extension is 523, and the IP address of my phone is 192.168.4.252. It stopped ringing after about 60 seconds.</div>
<div><br></div><div>Am I correct here? Is my phone doing it? If so, how do I change it? (I should probably ask that in the Polycom forum.)</div><div><br></div><div>-- </div><div>Steve</div><div class="gmail_extra"><br><br>
<div class="gmail_quote">On Mon, Mar 17, 2014 at 3:29 PM, Avi Marcus <span dir="ltr"><<a href="mailto:avi@avimarcus.net" target="_blank">avi@avimarcus.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<p>Usually the carrier lets it go as long as the remote end lets, afaik.<br>
Wiki says default timeout is 60 seconds: <a href="http://wiki.freeswitch.org/wiki/Variable_call_timeout" target="_blank">http://wiki.freeswitch.org/wiki/Variable_call_timeout</a></p>
<p>Try that and let us know the exact hangup if you still have a problem.</p><span class=""><font color="#888888">
<p>-Avi</p>
</font></span><div class="gmail_quote"><div><div class="h5">On Mar 18, 2014 12:06 AM, "Steven Schoch" <<a href="mailto:schoch%2Bfreeswitch.org@xwin32.com" target="_blank">schoch+freeswitch.org@xwin32.com</a>> wrote:<br type="attribution">
</div></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div><div class="h5">
<div dir="ltr">Today I had to call one of those old-school answering machines on a POTS line that had been turned off remotely, accidentally. To turn it back on, I had to call it and let the line ring 15 times (about one and a half minutes). But when I called through our FreeSWITCH PBX, the call was disconnected. Is this timeout in FreeSWITCH, or is my provider?<div>
<br></div><div>-- </div><div>Steve</div></div>
</div></div></blockquote></div></blockquote></div><br></div></div></div>