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Freeswitch-1.6.5 (video Conference)
by Veerabhadrarao Kankatala 25 Apr '16

25 Apr '16

Hello, -- I am currently working on video conference module in freeswitch-1.6.5 -- I am using verto client for confernece (i configured just 3 verto users in different hosts) and when i dial 3500 extension from the verto user, the conference is starting but the main problem is if 3 users connected with the 3500 conference, the audio is gone(too much noice, no audio clearence) and CPU usage is reached to 150%+ and i follow below configuration also ulimit -c unlimited # The maximum size of core files created. ulimit -d unlimited # The maximum size of a process's data segment. ulimit -f unlimited # The maximum size of files created by the shell (default option) ulimit -i unlimited # The maximum number of pending signals ulimit -n 999999 # The maximum number of open file descriptors. ulimit -q unlimited # The maximum POSIX message queue size ulimit -u unlimited # The maximum number of processes available to a single user. ulimit -v unlimited # The maximum amount of virtual memory available to the process. ulimit -x unlimited # ??? ulimit -s 240 # The maximum stack size ulimit -l unlimited # The maximum size that may be locked into memory. ulimit -a # All current limits are reported. i am using using Centos 6.5 with 4GB Ram. Could you any one help me to resolve this problem, i want to know following thigs --> Video confernece configuration --> Required system configuration --> and also i would like to know how many possible video conferences can be handle by freeswitch-1.6.5 Please help me. -- Thanks & Regards Veerabhadrarao Kankatala

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switch_rtp not changing IP with SRTP
by fs＠hellea.net 25 Apr '16

25 Apr '16

Hi, when calling to a linphone on a VPN without SRTP enabled on the leg-B, FS does [INFO] switch_rtp.c:6411 Auto Changing audio port from 192.168.1.35:7076 to 172.16.23.207:7076 But it doesn't when SRTP is enabled on the leg-B. How to force that behaviour when SRTP is enabled? Kind regards, JCh

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Don't report bugs or issues here please use jira.freeswitch.org
by Anthony Minessale 25 Apr '16

25 Apr '16

Friendly Reminder! -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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Verto registrations
by Gregor Nanger 25 Apr '16

25 Apr '16

Is there a way to flush specific verto registration. When I show status: verto status, somehow one registration is hanging. Can I flush specific verto registration from cli or ESL? When does session expire?

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Verto
by Gregor Nanger 25 Apr '16

25 Apr '16

I am building my web phone with verto from scratch and using this tutorial: http://evoluxbr.github.io/verto-docs/ BAsically everything is ok, I can register and make calls, but I cannot hear audio. Mic is working, but not speakers. I see error: Cannot read property 'sinkId' of undefined. This is error line: $.verto.dialog.prototype.setAudioPlaybackDevice = function(sinkId, callback, arg) { var dialog = this; var element = dialog.audioStream; * if (typeof element.sinkId !== 'undefined') {* I get dialog object without audioStream property and hence element is null. I am using self signed certificate. Could this be cause? Best regards, Gregor

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Extensions not registered on IPv6 address
by Shisheer Teli 25 Apr '16

25 Apr '16

Dear Team, I installed FreeSWITCH. Directory and dialplan are default ones. My phones are able to register when using just IPv4, and I can call from one to another. When I try the same thing with IPv6, phones not registered. Am I doing something wrong? Maybe I forgot to configure something, but I can't see what it is. -- Regards, Shisheer T

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Programmatic way to get BOTH Channel IDs of a bridged call?
by Chris Mandra 24 Apr '16

24 Apr '16

Hey guys - I have another questions: Is there a programatic way to obtain BOTH the channel ids of a bridged call? I only have the uuid of one call leg. How do i get the other uuid? (similar to what "show calls" or "show channels" shows in cli but I need to start with one call leg uuid and get the other from it) Thanks! chris

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Freeswitch sound installation is failing on CentOS 7
by Varsha Agarwal 23 Apr '16

23 Apr '16

Hi, I have installed FreeSwitch couple of times on CentOS with no problems. I am doing it on a fresh CentOS 7 VM and running into issues installing sounds. yum install sox freeswitch-sounds* This command fails with error 404 not found. I tried yum clean all and yum update but no luck. Can someone help? Thanks, Varsha

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Verto Caller Id and Caller name
by Gregor Nanger 23 Apr '16

23 Apr '16

I can set in verto javascript initialization caller id and caller name. But can I force or override caller id and name in user registration variables or somewhere else, because I do not want to set caller name in javascript, since user can tweak it.. Any experience?

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module not unloading
by Chris Mandra 23 Apr '16

23 Apr '16

Hi Guys - I hope you are well I never really resolved the issue I was having with modules saying they're unloading, but not really unloading (unless I restart freeswitch) I’m having that issue with the module not unloading again when I add in some flex/bison parser code. It seems the parser is causing the problem where module doesn’t unload. What's the best way to diagnose this issue. freeswitch reports module as unloading but lsof shows it is still being retained Thanks! chris

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