<p dir="ltr">Can you record this call with tcpdump?</p>
<div class="gmail_quote">16 февр. 2015 г. 18:55 пользователь "Charles Wang" <<a href="mailto:lazy.charles@gmail.com">lazy.charles@gmail.com</a>> написал:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi, Kamil,<div><br></div><div>All my clients(SIP) and server(with FXS/FXO) are in the same NAT environment that can access internet resources. And I start freeswitch with "freeswitch -nc". </div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-02-16 0:34 GMT+08:00 Kamil Nigmatullin <span dir="ltr"><<a href="mailto:kamil.nigmatullin@gmail.com" target="_blank">kamil.nigmatullin@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">Is your client behined NAT?</p>
<div class="gmail_quote">13 февр. 2015 г. 12:42 пользователь "Charles Wang" <<a href="mailto:lazy.charles@gmail.com" target="_blank">lazy.charles@gmail.com</a>> написал:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr"><span style="font-size:14px">Hi all,</span><div style="font-size:14px"><br></div><div style="font-size:14px">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.</div><div style="font-size:14px"><br></div><div style="font-size:14px">When I try to make call from FXO and it bridges to SIP device named 1234 via the following dialplan.</div><div style="font-size:14px"><div><br></div><div><div> <extension name="my call test"></div><div> <condition field="destination_number" expression="^(1234)$"></div><div> <action application="set" data="call_timeout=30"/></div><div> <action application="bridge" data="user/1234"/></div><div> </condition></div><div> </extension></div></div><div><br></div><div>Before SIP 1234 answers the call, the caller (FXO) hangup call before 1234 answered. But the callee (SIP 1234 or FXS) is still ringing and stop ring until call-timeout about 30 seconds.</div><div><br></div><div>It is the same condition if the callee is FXS device.<br></div><div><br></div><div>I can find two channels during ringing (A leg & B leg).</div><div>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 ).</div><div><br></div><div>There are two legs (A leg & B leg) after the caller FXO had hangup.</div><div><br></div><div><div>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</div><div>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</div><div>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</div></div><div><br></div><div><br></div><div><div>Can anyone help me to solve it or tell me why?</div><div><div><img src="https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif"></div></div></div></div><div><br></div>-- <br><div>Best Regards<br>Charles<br></div>
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