Hi Bote, I know the incoming call comes from VOIP Gateway that uses SIP protocol can work well in this scenario. And I don't think it is possible to ask customers to support extra cost in loop battery if I want FS to be a IP-PBX solution with FXS/FXO interfaces on it. Best regards, Charles 2015-02-18 4:41 GMT+08:00 Bote Man <bote_radio@botecomm.com>:
Detecting the loop battery interruption is the best way to do it if that is available on the analog line.
Bote
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Charles Wang *Sent:* Tuesday, 17 February, 2015 15:06 *To:* FreeSWITCH Users Help
*Subject:* Re: [Freeswitch-users] B leg ringing when Caller hangup before answer
Hi all,
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.
Best regards,
Charles
2015-02-18 3:48 GMT+08:00 Charles Wang <lazy.charles@gmail.com>:
Hi Bote,
I use analog card named TDM400P and there are two FXS and two FXO on it. You can find it from this url http://www.ryu.com.tw/image/A400P.jpg. There is a white power outlet behind it.
I am setting it at freetdm.conf and freetdm.conf.xml. But I don't know what loop battery is.
Is there any setting I can use when I try to make a call in <extensions> segement?
Thank you for your reply.
Best regards,
Charles
2015-02-17 16:00 GMT+08:00 Bote Man <bote_radio@botecomm.com>:
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.
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.
Bote
*From:* Charles Wang *Sent:* Thursday, 12 February, 2015 11:26 *Subject:* Re: [Freeswitch-users] B leg ringing when Caller hangup before answer
Hi Brian,
I ever try to answer the A leg in dialplan before bridge to B leg. But the condition is the same.
B leg is still ringing after I hangup the FXO.
I notice that the ringing of B leg will stop if the inbound tdm leg is FXS.
Let me know if you have any suggestion.
Best regards,
Charles
2015-02-12 23:43 GMT+08:00 Brian West <brian@freeswitch.org>:
You may wish to answer that inbound tdm leg before ringing out to the sip device.
On Thu, Feb 12, 2015 at 3:47 AM, Charles Wang <lazy.charles@gmail.com> wrote:
Hi all,
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.
When I try to make call from FXO and it bridges to SIP device named 1234 via the following dialplan.
<extension name="my call test">
<condition field="destination_number" expression="^(1234)$">
<action application="set" data="call_timeout=30"/>
<action application="bridge" data="user/1234"/>
</condition>
</extension>
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.
It is the same condition if the callee is FXS device.
I can find two channels during ringing (A leg & B leg).
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 ).
There are two legs (A leg & B leg) after the caller FXO had hangup.
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
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
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
Can anyone help me to solve it or tell me why?
--
Best Regards Charles
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--
Best Regards Charles
--
Best Regards Charles
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Best Regards Charles