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Call channel cannot cleaned normally
by 王聡 16 Nov '17

16 Nov '17

Hey all, I had made an ivr system based on lua and MySQL, and recently I found a call record which can’t be cleaned from my db. The freeswitch log had attached in the mail. I’m not sure if this call has any problem, but it lasted about 20min so I think it is a normal video call. I had tried uuid_kill but returned an error. Finally I had no choice but delete this record by sql command. Anyone had any idea about this situation? Regards.

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Re: [Freeswitch-users] CDRs and Bridge to multiple destination numbers
by itadebayo 16 Nov '17

16 Nov '17

Thanks. Ismail Sent from my Samsung Galaxy smartphone. -------- Original message --------From: Volkan Oransoy <voransoy(a)gmail.com> Date: 15/11/2017 7:35 p.m. (GMT+01:00) To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] CDRs and Bridge to multiple destination numbers Ismail, variable_last_bridge_to should contain uuid of the bridged channel. Cheers /Volkan > On 15 Nov 2017, at 17:24, Ismail Gmail <itadebayo(a)gmail.com> wrote: > > Hi all. > > I am bridging from a Lua script to multiple destination numbers using comma or pipe operator. How can I identify and record in the CDR the destination number that picks the call? > > Thanks. > > Ismail > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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mod_sms chat messages delivery notification
by Marko Djuric 15 Nov '17

15 Nov '17

Hi all, We're using mod_sms for SIP SIMPLE messaging with delivery notifications (message/imdn+xml). The Call-ID is changed in the chatplan, and this causes issues with matching the delivery notifications to the original message. Changing the Call-ID itself is not an issue, but I'm wondering if it is possible to read the original Call-ID and/or manually set the Call-ID in the chatplan. The message:getHeader("sip_call_id") does not contain the Call-ID. My understanding is that this is due to skip_global_process=false variable set by mod_sms. Setting the sip_call_id variable also does not produce a desired effect, as it is added as another header instead of setting the Call-ID one. The question is - how to match the messages before and after the chatplan? Any help is appreciated. Thank you! Marko

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flite Alternative
by Don Hawkins 15 Nov '17

15 Nov '17

Are their any good alternatives to Festival Lite TTS either paid or free? I'm looking to improve the quality...

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CDRs and Bridge to multiple destination numbers
by Ismail Gmail 15 Nov '17

15 Nov '17

Hi all. I am bridging from a Lua script to multiple destination numbers using comma or pipe operator. How can I identify and record in the CDR the destination number that picks the call? Thanks. Ismail

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Mod LUA memory leak
by Jai Rangi 15 Nov '17

15 Nov '17

I have been using LUA heavily and everything has been fine till FS 1.6.13. Since I upgraded one system to 1.6.17 and 1.6.19, I see this error just after 2-3 hours of FS restart. 2017-11-14 21:53:23.113817 [ERR] mod_lua.cpp:203 ...ocal/freeswitch/share/freeswitch/scripts/xml_handler.lua:100: LuaSQL: error connecting: Out of memory. stack traceback: [C]: in function 'assert' ...ocal/freeswitch/share/freeswitch/scripts/xml_handler.lua:100: in main chunk 2017-11-14 21:53:23.113817 [ERR] mod_lua.cpp:271 LUA script parse/execute error! Same configuration, same modules, same scripts, same library. Everything works fine till 1.6.13. Wondering if anyone experiencing something similar. I can open a bug report, but just wanted to make sure this is not only me. Thank you, -Jai

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Video recording
by Freeswitch user 15 Nov '17

15 Nov '17

Hello users, Is there any way in freeswitch to record two way video calls. Please suggest!! Thanks

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CDRs and continue_on_fail
by Gregor Nanger 15 Nov '17

15 Nov '17

Hi! I have simple scenario to bridge call to gateway and use fallback gateway if something is wrong with first one. So this is short dialplan: continue_on_fail=NO_ROUTE_TRANSIT_NET, ........ bridge to gw1 bridge to gw2 But if gw1 fails and call is bridged through gw2, I do not get CDR from gw1. Is there a way to tell FS to post also CDR from first bridge as flow section of CDR or separate post to server? Thank you for your help.

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uuid_fileman
by Mirko Brankovic 15 Nov '17

15 Nov '17

Hi, What would be the proper way to execute fileman api from custom plugin? Is switch_api_execute(fileman, command, ...stream) good solution? I don't seem to get response I=back in stream,... probably doing something wrong. any suggestion is helpful, Thanks, Mirko

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Record by mod_fsv
by 王聡 15 Nov '17

15 Nov '17

Hey all, I had got an error on video call recording by mod_fsv. Here are parts of logs on fs_cli: EXECUTE sofia/external/1004(a)192.168.109.2 record_session(/tmp/test.fsv) 2017-11-15 19:19:50.909190 [INFO] mod_fsv.c:837 Opening File [/tmp/test.fsv] 8000hz … 2017-11-15 19:19:50.939189 [NOTICE] switch_core_io.c:1202 Activating write resampler 2017-11-15 19:19:50.949188 [DEBUG] switch_rtp.c:7308 Correct audio ip/port confirmed. 2017-11-15 19:19:50.949188 [DEBUG] switch_core_io.c:448 Setting BUG Codec PCMU:0 2017-11-15 19:19:50.949188 [NOTICE] switch_core_io.c:1202 Activating write resampler 2017-11-15 19:19:50.989187 [DEBUG] sofia.c:7084 Channel sofia/external/1004(a)192.168.109.2 entering state [ready][200] 2017-11-15 19:19:51.039187 [DEBUG] switch_rtp.c:6758 Correct video RTCP ip/port confirmed. 2017-11-15 19:19:51.059187 [DEBUG] switch_rtp.c:7308 Correct audio ip/port confirmed. 2017-11-15 19:19:51.189189 [DEBUG] switch_rtp.c:6758 Correct video RTCP ip/port confirmed. 2017-11-15 19:19:51.449187 [ERR] mod_fsv.c:969 You are asking to write 4096 bytes of data which is not supported. Please set enable_file_write_buffering=false to use .fsv format 2017-11-15 19:19:51.449187 [ERR] switch_ivr_async.c:1169 Error writing /tmp/test.fsv The dialplan looks like this: <action application="export" data="nolocal:absolute_codec_string=OPUS,VP8"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="record_concat_video=true"/> <action application="set" data="enable_file_write_buffering=false"/> <action application="set" data="execute_on_answer=record_session /tmp/test.fsv”/> The call itself has no problem, and I got a test.fsv file after hangup but the size is only 316 Byte. I had tried uuid_record and got the same result. Anyone has any idea? PS. Both mod_av and mod_vlc cannot work properly, I had no choice but try mod_fsv. Server OS: Debian 8.5 Jessie FS version: FreeSWITCH Version 1.6.19+git~20171025T192126Z~86184a2f73~64bit (git 86184a2 2017-10-25 19:21:26Z 64bit) Regards.

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