<div>Ok, as discussed with Tony on IRC channel I followed his directions which lead to a successfull outcome (like it always does I might add :).</div>
<div> </div>
<div>One has to use tone_detect app in FreeSWITCH dialplan in order to check for busy tones coming from the PSTN side and if matched fire a hangup application. This is the snippet of my test dp that does the trick (from extension Local_extensions in default.xml):</div>
<div> </div>
<div><anti-action application="tone_detect" data="busy 425,0 r +100 hangup 16 4"/><br><anti-action application="bridge" data="<a href="mailto:user/$%7Bdialed_extension%7D@$%7Bdomain_name%7D%22/">user/${dialed_extension}@${domain_name}"/</a>><br>
</div>
<div>This means that FS will listen to freq of 425 Hz and wait for 4 positive detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 Hz is the freq telco here uses; for other countries I suggest getting the ITU world tones pdf file and check there):</div>
<div> </div>
<div>2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 1/4<br>2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 2/4<br>2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 3/4<br>
2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback() TONE busy HIT 4/4<br>2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback() TONE busy DETECTED<br></div>
<div>2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING]</div>
<div> </div>
<div>Regards,</div>
<div>Ognjen</div>
<div> </div>
<div class="gmail_quote">On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija <span dir="ltr"><<a href="mailto:oseslija@gmail.com">oseslija@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div>I tried similar setup with my analog card (X100P) and I'm having same issue. Call is not hungup on the oz side once the caller ends. My telco doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck to detecting busy tone from the telco side. I'll try to modify tones.conf accordingly.</div>
<div> </div>
<div>Regards,</div>
<div>Ognjen<br>(sekil)<br></div>
<div>
<div></div>
<div class="Wj3C7c">
<div class="gmail_quote">On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">This is a common issue with analog phones even traditional answering machines suffer from it.<br>I'm sure you must have had an answering machine at some point that has dial tone as the message it receives.<br>
<br>Unless FreeSWITCH has some hint that the call has hungup it will not stop trying to complete the call.<br><br>If the other side is sending a busy tone to indicate hangup it's possible to use the tone_detect app to pick<br>
up on the tones and abort the call.<br><br>Another thing you could do if you have unlimited inbound is explicitly answer the call in the dialplan before<br>you call your sip phones this will give you a more profound hangup detection but it will make every call count<br>
even when nobody answers.<br><br><br><br>
<div class="gmail_quote">
<div>
<div></div>
<div>On Tue, Jan 20, 2009 at 10:46 AM, Tomás <span dir="ltr"><<a href="mailto:tomasborrella@gmail.com" target="_blank">tomasborrella@gmail.com</a>></span> wrote:<br></div></div>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<div>
<div></div>
<div>Hi all,<br><br>I'm configuring my home PBX using FreeSwitch. I'm using a X101P card configured as FXO (conected to analog PSTN line) and I have several IP phones and softphones conected to FreeSwitch.<br><br>
I can call from an IP phone to other IP phone (the same with the softphones) and also from an IP phone (or softphone) to an external number thought PSTN.<br><br>When I call from an external analog phone to FreeSwitch, I bridge the call to all internal IP phones and softphones and they ring, but the problem is that when I hang up the call in the external phone, all internal phones (IP phones and softphones) keeps ringing...<br>
<br>I'm pretty sure the problem is that FreeSwitch don't receive the hang up, because I cann't see anything on the log.<br><br>I've also created my own tones.conf for my country (Spain) but I'm not sure if it's ok (but I have the same problem with hang up)<br>
<br>I've googled the list, and I've found several people with a similar problem but no solution...<br><br>That's my pastebin with the most importants printouts and config files:<br><a href="http://pastebin.freeswitch.org/6822" target="_blank">http://pastebin.freeswitch.org/6822</a><br>
<br>Thank you very much in advance.<br><br></div></div>
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<br></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
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