<div dir="ltr"><div>Hi all,</div><div><br></div>I ever try to record the channel of FXO with dahdi_monitor. I can find ring tone and ringback tone in rx and tx when FXO caller is calling FXS or SIP callee. But there is no busy tone when caller hangup the call BEFORE the callee(FXS/SIP) answers it or the call-timeout condition is reached. I think it is correct, isn't it? If it is correct, I can't use busy_detect to handle the disconnect of channels.<div><br></div><div>Best regards,</div><div>Charles</div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-02-18 3:48 GMT+08:00 Charles Wang <span dir="ltr"><<a href="mailto:lazy.charles@gmail.com" target="_blank">lazy.charles@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi Bote,<div><br></div><div>I use analog card named TDM400P and there are two FXS and two FXO on it. You can find it from this url <a href="http://www.ryu.com.tw/image/A400P.jpg" target="_blank">http://www.ryu.com.tw/image/A400P.jpg</a>. There is a white power outlet behind it.</div><div><br></div><div>I am setting it at freetdm.conf and freetdm.conf.xml. But I don't know what <span style="color:rgb(31,73,125);font-family:Calibri,sans-serif;font-size:14.6666669845581px">loop battery is.</span></div><div><span style="color:rgb(31,73,125);font-family:Calibri,sans-serif;font-size:14.6666669845581px">Is there any setting I can use when I try to make a call in <extensions> segement?</span></div><div><span style="color:rgb(31,73,125);font-family:Calibri,sans-serif;font-size:14.6666669845581px"><br></span></div><div><span style="color:rgb(31,73,125);font-family:Calibri,sans-serif;font-size:14.6666669845581px">Thank you for your reply.</span></div><div><span style="color:rgb(31,73,125);font-family:Calibri,sans-serif;font-size:14.6666669845581px"><br></span></div><div><span style="color:rgb(31,73,125);font-family:Calibri,sans-serif;font-size:14.6666669845581px">Best regards,</span></div><div><span style="color:rgb(31,73,125);font-family:Calibri,sans-serif;font-size:14.6666669845581px">Charles</span></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="h5">2015-02-17 16:00 GMT+08:00 Bote Man <span dir="ltr"><<a href="mailto:bote_radio@botecomm.com" target="_blank">bote_radio@botecomm.com</a>></span>:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5"><div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">It sounds like your originating caller is analog and is not providing supervision signal to indicate to the FXO port that it has released the call, therefore FS continues as if the call is still held active. <u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">In some configurations the loop battery is interrupted briefly to indicate that the call has been released. In other cases you must detect call progress tones that indicate that the caller is no longer present. In some cases there is no indication provided to the FXO port and only timers and the Leg B will help you.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Bote<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> <u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> Charles Wang<br><b>Sent:</b> Thursday, 12 February, 2015 11:26<br><b>Subject:</b> Re: [Freeswitch-users] B leg ringing when Caller hangup before answer<u></u><u></u></span></p><div><div><p class="MsoNormal"><u></u> <u></u></p><div><div><p class="MsoNormal">Hi Brian,<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><p class="MsoNormal">I ever try to answer the A leg in dialplan before bridge to B leg. But the condition is the same.<u></u><u></u></p><div><p class="MsoNormal">B leg is still ringing after I hangup the FXO.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I notice that the ringing of B leg will stop if the inbound tdm leg is FXS.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Let me know if you have any suggestion.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Best regards,<u></u><u></u></p></div><div><p class="MsoNormal">Charles<u></u><u></u></p></div></div><div><p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">2015-02-12 23:43 GMT+08:00 Brian West <<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>>:<u></u><u></u></p><div><p class="MsoNormal">You may wish to answer that inbound tdm leg before ringing out to the sip device.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p><div><div><div><p class="MsoNormal">On Thu, Feb 12, 2015 at 3:47 AM, Charles Wang <<a href="mailto:lazy.charles@gmail.com" target="_blank">lazy.charles@gmail.com</a>> wrote:<u></u><u></u></p></div></div><blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in"><div><div><div><p class="MsoNormal">Hi all,<u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I have a server with freeswitch 1.4.15 + freetdm(FXS/FXO). I think there is a bug in inbound call via the freetdm FXO device.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">When I try to make call from FXO and it bridges to SIP device named 1234 via the following dialplan.<u></u><u></u></p></div><div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><div><p class="MsoNormal"> <extension name="my call test"><u></u><u></u></p></div><div><p class="MsoNormal"> <condition field="destination_number" expression="^(1234)$"><u></u><u></u></p></div><div><p class="MsoNormal"> <action application="set" data="call_timeout=30"/><u></u><u></u></p></div><div><p class="MsoNormal"> <action application="bridge" data="user/1234"/><u></u><u></u></p></div><div><p class="MsoNormal"> </condition><u></u><u></u></p></div><div><p class="MsoNormal"> </extension><u></u><u></u></p></div></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Before SIP 1234 answers the call, the caller (FXO) hangup call before 1234 answered. But the callee (SIP 1234) is still ringing and stop ring after about 30 seconds.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">It is the same condition if the callee is FXS device.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I can find two channels during ringing (A leg & B leg).<u></u><u></u></p></div><div><p class="MsoNormal">After the caller(FXO) onhook, the A leg is still alive and A leg will not be hangup before the callee (FXS or SIP) stop ring ( call-timeout ).<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">There are two legs (A leg & B leg) after the caller FXO had hangup.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><div><p class="MsoNormal">uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context<u></u><u></u></p></div><div><p class="MsoNormal">53ce4fe9-9511-4625-87ae-1448421c9810,inbound,2015-02-12 17:33:11,1423733591,FreeTDM/2:2/1234,CS_EXECUTE,unknown,unknown,,1234,bridge,freetdm/FXS1/1,XML,TEST,PCMU,8000,64000,PCMU,8000,64000,,charles,,,RINGING,,,,,,,unknown,unknown,,1234,XML,default<u></u><u></u></p></div><div><p class="MsoNormal">cf5a7ee4-cbfd-48e4-ab4e-4d757216712c,outbound,2015-02-12 17:33:12,1423733592,FreeTDM/1:1/,CS_CONSUME_MEDIA,unknown,unknown,,1,,,XML,default,,,,,,,,charles,,,RINGING,Outbound Call,1,,53ce4fe9-9511-4625-87ae-1448421c9810,,,unknown,unknown,,1,XML,default<u></u><u></u></p></div></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><div><p class="MsoNormal">Can anyone help me to solve it or tell me why?<u></u><u></u></p></div><div><p class="MsoNormal"><span style="color:#888888"><u></u> <u></u></span></p></div></div><div><p class="MsoNormal"><span style="color:#888888"><u></u> <u></u></span></p></div><p class="MsoNormal"><span style="color:#888888">-- <u></u><u></u></span></p><div><p class="MsoNormal"><span style="color:#888888">Best Regards<br>Charles<u></u><u></u></span></p></div></div></div><p class="MsoNormal"><u></u> <u></u></p></div></div></blockquote></div></div></div></div></div></div></div></div><br></div></div><span class="">_________________________________________________________________________<br>
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