B leg ringing when Caller hangup before answer
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
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
_________________________________________________________________________ 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
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-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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
_________________________________________________________________________ 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
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ 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
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
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
_________________________________________________________________________ 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
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
_________________________________________________________________________ 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
-- Best Regards Charles
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 _________________________________________________________________________ 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 -- Best Regards Charles
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
_________________________________________________________________________ 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
--
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
Your FXO port connects to an analog telephone line that supplies battery FROM THE OTHER END, typically a central office switch or PBX. This is how telephones have worked for 100 years; there is no extra cost to it. Because it is so common I suggested this as your first method to try since it works well, all you must do is tell your TDM400P to watch for it. (This is a concept similar to the gentleman who is receiving invalid Caller*ID info from the far end switch; it is out of his control and the other end must correct it.) If that is not available, then try detecting call progress tones coming from the far end of the FXO connection. If this can’t be done, then there is nothing left to do. In any case, this has NOTHING to do with SIP. You must detect conditions on the analog line in the analog domain using the equipment connected to that analog telephone line. That is the source of the information that you seek, not in FreeSWITCH and not in SIP. Bote From: Charles Wang Sent: Thursday, 19 February, 2015 13:28 Subject: Re: [Freeswitch-users] B leg ringing when Caller hangup before answer 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 _________________________________________________________________________ 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 -- 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
The FXO port connects to an analog telephone line. The analog phone uses the electrical current on the phone line to operate. 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. 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. 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. This is why analog telephony is no fun compared to VoIP L 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 14:48 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] B leg ringing when Caller hangup before answer 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 _________________________________________________________________________ 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
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@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.
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.
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.
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.
This is why analog telephony is no fun compared to VoIP L
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 14:48 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] B leg ringing when Caller hangup before answer
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
_________________________________________________________________________ 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
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--
Best Regards Charles
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---------------------------------------------------- _________________________________________________________________________ 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.com
I have used the Patton FXO AND FXS gateways to bring analog lines into FreeSWITCH with success, but their programming requires a 4 year college degree to figure out! The Patton gateways will detect call progress tones as well as loop battery interrupt to know that the FXO has released the call and will process that correctly to tell FS to end the call. Thanks for your experience as switching to something else might be better, instead of testing different scenarios. Bote -----Original Message----- From: covici@ccs.covici.com Sent: Tuesday, 17 February, 2015 15:19 Subject: Re: [Freeswitch-users] B leg ringing when Caller hangup before answer 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.
From: Charles Wang Sent: Tuesday, 17 February, 2015 14:48 Subject: Re: [Freeswitch-users] B leg ringing when Caller hangup before answer
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.
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_ad dr,dest,application,application_data,dialplan,context,read_codec,read_ rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostna me,presence_id,presence_data,callstate,callee_name,callee_num,callee_d irection,call_uuid,sent_callee_name,sent_callee_num,initial_cid_name,i nitial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_c ontext
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,,charle s,,,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
______________________________________________________________________ ___ 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-use rs http://www.freeswitch.org
--
Best Regards Charles
---------------------------------------------------- Alternatives:
---------------------------------------------------- ______________________________________________________________________ ___ 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-use rs http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.com _________________________________________________________________________ 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
participants (4)
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Bote Man
-
Brian West
-
Charles Wang
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covici＠ccs.covici.com