April 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

More pixelated video canvas on larger number of users
by Chad Phillips 05 May '17

05 May '17

Running Verto/mod_conference videoconference on 1.6.11, I’ve noticed that the entire canvas resolution seems more pixelated when the number of users connected to the videoconference goes up. If just one person is connected, the image is very consistent and clear, but getting into the 7-10 person range, the quality drops noticeably. And I’m not talking about the quality of one particular video on the canvas, but the entire canvas quality. I’ve done a recording of a videoconference on the server with a larger number of users, and the video quality there is clear and consistent, so it doesn’t seem to be an issue with either receiving or muxing the feeds, but in how the end user is receiving the muxed video. I tried playing with the the ‘video-quality’ and ‘video-codec-bandwidth’ conference params — increasing the video-quality from 1 to 3 didn’t seem to have much of an impact, increasing the video-codec-bandwidth from 1mb to 2mb used quite a bit more CPU, but didn’t seem to positively impact the video quality, either. Curious if I’m missing something in the config, or if there’s something else I can do to improve the video quality with a larger number of users. Chad

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Changing mode for CDRs
by Rick Jarvis 04 May '17

04 May '17

How do I set the permissions for the CDR files that FreeSWITCH writes? There don’t appear to be any settings that I can find apart from in config.FS0, although changing in this file doesn’t make any difference (and it doesn’t appear to be in typical FS format, I’m not actually sure what it does?!)… Thanks R

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IP PBX selection
by Roamer2998 03 May '17

03 May '17

Hi all, I'm new to VoIP, now we have a project that needs a PBX with client APPs. In our team we have argument for choosing PBX. By so far, we have following candidates: A: Open source 1) Asterisk PBX (http://www.asterisk.org ) (with longest history that almost every one knows it) 2) FreeSwitch (http://www.freeswitch.org ) (A lot people recommended it to us) B: Commercial 1) Vodia PBX (http://www.vodia.com) It comes from SNOM, but acquired by a HongKong company now 2) PortSIP PBX (http://www.portsip.com/portsip-pbx ). It also includes VoIP SDK, WebRTC and offer rebranding app for free. My boss prefers the Open Source PBX since they are free, but our CTO prefers the commercial editions, according to whom the business PBX has better support, and the performance is good, and easy to use - considering our team all are new to VoIP/PBX. We have did some searching of FreeSwitch, it seems the sofia stack comes from Nodia. Here's my question: 1. Does the Nokia still maintain this stack or not? 2. Any commercial support by FreeSwitch developers? Can I got the quotation? 3. Is it easy to compile and setup FreeSwitch? 4. Which FreeSwitch version is recommended? ( 1.6?) Thanks in advance . Best regards,

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FS core dumps
by Rahul MathuR 01 May '17

01 May '17

Hello guys, I'm trying to Auth over RADIUS from inside Lua script - session:execute("auth_function","in "..cli..", in "..dnid..", out AUTH_RESULT"); But FS keeps on dumping core files - SIGSEGV inside "extract_out_variable" inside mod_rad_auth.so My guess is that I have not defined AUTH_RESULT anywhere in the script. But then again I haven't find any example on this either. Any help would be much appreciated. -- Warm Regds. MathuRahul

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Any plan to support DPDK?
by Raymond 30 Apr '17

30 Apr '17

Hi, guys Is there's any plan to develop a "DPDK Freeswitch version" ? DPDK is really a big improve of network . Raymond

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答复: freeswitch crash when using record_session in dialplan
by Harry.Z 30 Apr '17

30 Apr '17

Dear Raymond, Are you Chinese? I attach the network topology, dialplan and console log. I make a lot of tests. The FS always crash (probability 70%) if I enable the record_session in dialplan when I using the originate make the call. So I guess the record_session is the point. But the FS crash (probability 5%) if I disable the record_session in dialplan when I using the originate make the call. Now I delete <param name="apply-nat-acl" value="nat.auto"/> in sip profile internel.xml. It’s not happen again. But I’m not sure this issue resolved. Maybe it’s a bug on nat? --------------------------------- 附件是网络拓扑、dialplan和console日志。我做了大量的测试。如果话机和FS不在同一个网段，用originate这个命令发起呼叫FS就会崩溃，但过一段时间又是正常的，这期间没有做任何的修改。如果把record_session删掉，崩溃的机率就非常小，估计都不到5%。我刚才试着把internal profile中的<param name="apply-nat-acl" value="nat.auto"/>这一行删掉，到目前为止是正常的，但我还不确定是不是这个引起的。我怀疑这是一个Bug。 Harry 发件人: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] 代表 Raymond 发送时间: 2017年4月29日 13:20 收件人: FreeSWITCH Users Help 主题: Re: [Freeswitch-users] freeswitch crash when using record_session in dialplan attach your console log info please. And also your dialplan. ------------- 把你执行命令的时候，console 窗口的log信息贴上来。另外附上你的那个有 record_session 的 dialplan 。 Raymond

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freeswitch crash when using record_session in dialplan
by Harry.Z 29 Apr '17

29 Apr '17

Dear All, I'm new on FreeSWITCH. I installed a FreeSWITCH on Debian 8 by apt-get. freeswitch@FS> version FreeSWITCH Version 1.6.17-34-0fc0946~64bit (-34-0fc0946 64bit) I enabled record_session in dialplan. I attach the dialplan. I using originate command in fs_cli. freeswitch@FS> originate user/1001 1002 The Freeswitch will crash when 1001 answer. I didn't find any useful information in /var/log/freeswitch/freeswitch.log Am I missing something? Thanks a lot!!!! Harry

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答复: freeswitch crash when using record_session in dialplan
by Harry.Z 29 Apr '17

29 Apr '17

Dear Brian, Sorry, I miss a very important information. If the FreeSWITCH and the phone in the same subnet (10.5.15.0/24) is working fine. But if the FreeSWITCH and the phone in the different subnet (10.5.15.0/24 and 10.5.13.0/24) will crash. Harry 发件人: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] 代表 Brian West 发送时间: 2017年4月29日 0:13 收件人: FreeSWITCH Users Help 主题: Re: [Freeswitch-users] freeswitch crash when using record_session in dialplan I'm testing on v1.6 branch, can't replicate your issue, Please file the JIRA with the backtrace and log files. /b On Fri, Apr 28, 2017 at 10:49 AM, Michael Jerris <mike(a)jerris.com<mailto:mike@jerris.com>> wrote: https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA https://freeswitch.org/confluence/display/FREESWITCH/Debugging#Debugging-Ad… On Apr 28, 2017, at 5:35 AM, Harry.Z <Harry.Z(a)iv66.net<mailto:Harry.Z@iv66.net>> wrote: Dear All, I’m new on FreeSWITCH. I installed a FreeSWITCH on Debian 8 by apt-get. freeswitch@FS> version FreeSWITCH Version 1.6.17-34-0fc0946~64bit (-34-0fc0946 64bit) I enabled record_session in dialplan. I attach the dialplan. I using originate command in fs_cli. freeswitch@FS> originate user/1001 1002 The Freeswitch will crash when 1001 answer. I didn’t find any useful information in /var/log/freeswitch/freeswitch.log Am I missing something? Thanks a lot!!!! Harry _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto: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(a)lists.freeswitch.org<mailto: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(a)freeswitch.org<mailto:brian@freeswitch.org> Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST)<https://freeswitch.com/appointment> Allison prompts for FreeSWITCH: https://www.gofundme.com/allison-prompts-for-freeswitch Got Bugs? Report them here<https://freeswitch.org/jira>! | Reddit: /r/freeswitch<https://www.reddit.com/r/freeswitch> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest

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skip busy agent
by Robert Oldham 28 Apr '17

28 Apr '17

Is there a way to implement skip busy agent in freeswitch? I'm trying to send an incoming call to a ring group, without ringing anyone that is already on the phone or DND. Is there a simple way to do this? Thanks! Robert Oldham Oldham Technology (801) 877-2190 https://www.oldhamtechnology.com/

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Error 503 at startup
by William Colls 28 Apr '17

28 Apr '17

FreeSWITCH version 1.6.12 Debian Jessie 64 bit When the systen boots up, freesitch is started, but it does not connect to my voip provider. The following messages are reported in fs_cli: [NOTICE] sofia_reg.c:448 Registering voipms_demo [ERR} sofia_reg.c:2446 voipms_demo Failed Registration with status DNS Error [503] If I do a systemctl restart freeswitch.service, the connection is made without problem. Should I be talking to my voip provider about this, or is it something miss-configured within FreeSWitch? Thanks for your time William.

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