March 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Question on httapi
by Jai Rangi 09 Mar '17

09 Mar '17

Anyone using httapi module. There is a setting in httapi.conf <param name="file-cache-ttl" value="300"/> Is it possible to change that to different values in the dialplan dynamically. For example for one file.mp3 we want to 500 seconds and for another file.mp3 we dont want to cache at all. Thank you, Jai

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Streaming video conferences
by Volodymyr Fedorov 08 Mar '17

08 Mar '17

Hi did somebody using mod_vlc for streaming ? I tried to use following settings in conferences.conf.xml <param name="min-required-recording-participants" value="1"/> <param name="auto-record" value="vlc://#standard{access=http,mux=raw,dst= 192.168.1.8:8080/stream}"/> And get following in console: === 2017-03-08 23:55:17.622992 [INFO] mod_conference.c:370 Auto recording file: vlc://#standard{access=http,mux=raw,dst=192.168.1.8:8080/stream} 2017-03-08 23:55:17.622992 [ERR] switch_core_file.c:264 Invalid file format [vlc] for [#standard{access=http,mux=raw,dst=192.168.1.8:8080/stream}]! 2017-03-08 23:55:17.622992 [ERR] conference_record.c:271 Error Opening File [{channels=2,samplerate=48000,vw=1280,vh=720,fps=30.00}vlc://#standard{access=http,mux=raw,dst= 192.168.1.8:8080/stream}] 2017-03-08 23:55:17.622992 [ERR] switch_core_timer.c:117 Timer is not properly configured. === I`m running FreeSWITCH (Version 1.6.15 -32-bec4538 64bit) from packages on Debian 8. Maybe somebody may share working config section ? Thanks! -- Best regards, Volodymyr

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continue not working
by David Villasmil 08 Mar '17

08 Mar '17

Hello guys, Does anyone see anything wrong with this? The first extension gets executed, but not the second, even though i have continue="true" <extension name="do_this_for_all_9" continue="true"> <condition field="destination_number" expression="^(9.*)$"> <action application="set" data="rand_val=${expr(randomize(&x);ceil(random(0,10,&x)))}"/> <action application="log" data="ERR Random Value: <${rand_val}>"/> </condition> </extension> <extension name="local_call2"> <condition field="destination_number" expression="^(9.*)$"> <condition field="${cond(${rand_val} > 4 ? YES : NO)}" expression="^YES$"> <action application="set" data="continue_on_fail=true"/> <action application="ring_ready"/> <action application="sleep" data="3000"/> <action application="hangup" data="USER_BUSY"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="ring_ready"/> <anti-action application="sleep" data="3000"/> <anti-action application="sched_hangup" data="+10 allotted_timeout"/> <anti-action application="answer"/> <acti-action application="callcenter" data="support@default "/> <anti-action application="hangup"/> </condition> </condition> </extension> Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Memory leakage issue
by Siju Nair 08 Mar '17

08 Mar '17

Hi team, My FS server got hung, under server logs due freeswitch server got hung up. Below is the log i found.. echo 0 > /proc/sys/kernal/hung_task_timeout_secs"disables this message info: task freeswitch: 3034 blocked for more than 120 secs. and then freeswitch > waiting for background process to be ready. Please help me on the same. Sent from my iPhone

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Setting user to user information (UUI)
by Varsha Agarwal 07 Mar '17

07 Mar '17

Hi, In Freeswitch, is it possible to set SIP UUI (user to user information) before transferring the call another application? We want to add some information to the call so the application on the other end can retrieve it? Thanks, Varsha

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Packet Loss and Round Trip Time
by Oleg Stolyar 07 Mar '17

07 Mar '17

Hi guys, is there a way to get Packet Loss and Round Trip Time in FS CDR without using a 3rd party tool like VoIP Monitor?

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how to debug call (audio/video) quality
by Mundkowsky, Robert 07 Mar '17

07 Mar '17

We use FreeSWITCH for Verto video conference calls. Sometimes some users experience noise audio and video. How do I go about debugging call quality problems? >From what I have read, problems are likely due to packet loss and jitters, but I don' have a way to determine if this is the problem or a system issue. >From the recent FreeSWITCH Book, seems like using the HOMER might help. I have watched the Chrome webrtc charts before and I don't seem to find them useful, granted maybe I do not understand them well. Also if to packet loss and jitters is the problem, is there any way to try to fix such problems? Robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________

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playback PLS, M3U or XSPF file formats
by Anonim Stefan 07 Mar '17

07 Mar '17

Hi, Is there a way to playback playback PLS, M3U or XSPF shoutcast streams (e.g. [1])? I tried both mod_vlc[2] and mod_shout[3] without success. I tried playback a local pls file without success. [1] http://yp.shoutcast.com/sbin/tunein-station.pls?id=403432 [2] vlc://http://yp.shoutcast.com/sbin/tunein-station.pls?id=403432 <http://yp.shoutcast.com/sbin/tunein-station.pls?id=403432>[3] shout:// yp.shoutcast.com/sbin/tunein-station.pls?id=403432 Thank you, Stefan

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set_profile_var question
by Abaci B 07 Mar '17

07 Mar '17

Just wondering if there is something like set_profile_var to get a profile variable , there doesn't seem to be a get_profile_var or at least a way I can get the value of Caller-Orig-Caller-ID-Name Thanks

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FS account got hacked **urgent**
by Siju Nair 06 Mar '17

06 Mar '17

Hi Team my account got hacked and attacked using my DID number as caller id and making calls via my FS server. in logs i could notice this sofia/external/'hi'or'x'='x' ... what does this mean and how can they set my did as caller id and make calls... Urgent help needed. Thanks, Siju Nair

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