[Freeswitch-users] B leg ringing when Caller hangup before answer
covici at ccs.covici.com
covici at ccs.covici.com
Tue Feb 17 23:19:23 MSK 2015
I tried that card with freetdm and had very bad luck with it -- you
would be better served by an interface device such as Linksys or
Audiocode or some others to convert to sip.
Bote Man <bote_radio at botecomm.com> wrote:
> The FXO port connects to an analog telephone line. The analog phone uses the electrical current on the phone line to operate.
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> The central office or equipment at the OTHER end of that line provides the battery which is detected by your TDM400P. If that far end office does not interrupt the loop current on the analog telephone line, your FXO port has no way to detect that the caller at the far end has released the call, so FS continues the call. This is out of your direct control and is provided by the operator of the equipment at the far end.
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> There are crude ways to test this with an analog telephone connected to this analog line if the phone is powered by the battery on the line. I have tested the loop interrupt by blowing into the mouthpiece while the far end hangs up and listening for the sidetone (my sound returning to my own ear piece) to be interrupted briefly. Or you can use a voltmeter with fast response time to measure the voltage on the line while the far end hangs up the call.
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> If your tests reveal that the far end equipment is not interrupting the voltage on the phone line, then you must find another way to detect that the far end has released the call, typically by detecting call progress tones such as busy or reorder. Either way, the TDM400P must be able to detect this and translate it to a useful action on the SIP side.
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> This is why analog telephony is no fun compared to VoIP L
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> Bote
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> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Charles Wang
> Sent: Tuesday, 17 February, 2015 14:48
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] B leg ringing when Caller hangup before answer
>
>
>
> Hi Bote,
>
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> 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.
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> I am setting it at freetdm.conf and freetdm.conf.xml. But I don't know what loop battery is.
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> Is there any setting I can use when I try to make a call in <extensions> segement?
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> Thank you for your reply.
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> Best regards,
>
> Charles
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> 2015-02-17 16:00 GMT+08:00 Bote Man <bote_radio at 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.
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> 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.
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> Bote
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>
> 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.
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> B leg is still ringing after I hangup the FXO.
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> I notice that the ringing of B leg will stop if the inbound tdm leg is FXS.
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> Let me know if you have any suggestion.
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> Best regards,
>
> Charles
>
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> 2015-02-12 23:43 GMT+08:00 Brian West <brian at freeswitch.org>:
>
> You may wish to answer that inbound tdm leg before ringing out to the sip device.
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> On Thu, Feb 12, 2015 at 3:47 AM, Charles Wang <lazy.charles at gmail.com> wrote:
>
> Hi all,
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>
>
> 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.
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> When I try to make call from FXO and it bridges to SIP device named 1234 via the following dialplan.
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> <extension name="my call test">
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> <condition field="destination_number" expression="^(1234)$">
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> <action application="set" data="call_timeout=30"/>
>
> <action application="bridge" data="user/1234"/>
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> </condition>
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> </extension>
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> 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.
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> It is the same condition if the callee is FXS device.
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> I can find two channels during ringing (A leg & B leg).
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> 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 ).
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> There are two legs (A leg & B leg) after the caller FXO had hangup.
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> 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
>
>
>
>
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> Can anyone help me to solve it or tell me why?
>
>
>
>
>
> --
>
> Best Regards
> Charles
>
>
>
>
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>
> --
>
> Best Regards
> Charles
>
>
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--
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covici at ccs.covici.com
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