October 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Memory leak
by Sergey Safarov 30 Dec '16

30 Dec '16

I use monitoring tool (zabbix) to check health of FreeSwitch servers. I has found that after 60 days of uptime FS host has less free memory than after server reboot. This memory is returned back after FS daemon restart. On FS host is installed 66fe6e5. How to correctly detect memory leak in FS?

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March 23rd Cluecon Weekly lost?
by Tom Lynn 17 Dec '16

17 Dec '16

I've tried to play back the March 23rd entry of Cluecon Weekly on Youtube.com but keep getting "An error Occurred. Please try again later." Can this episode be reposted? or is it lost? Thank you. Tom

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Ubuntu 14.x and 16.x Packages
by Brian West 13 Dec '16

13 Dec '16

FreeSWITCHers, We have published both Ubuntu repositories: https://freeswitch.org/confluence/display/FREESWITCH/Ubuntu+16.04+Xenial https://freeswitch.org/confluence/display/FREESWITCH/Ubuntu+14.04+Trusty Please review and give us feedback. Also if you're new check out my signature for a discount code for the ebooks, This code applies to FreeSWITCH Cookbook And the new Mastering FreeSWITCH book. If you've not done so check them out, The entire team put in a lot of work writing and proofing the books. It will help you get started with FreeSWITCH faster. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu 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) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Enable H264 Video
by Jürgen Wendler 07 Nov '16

07 Nov '16

Hello everyone, i've found some amazaing articles about FS and so i thought i can give it a try. So I followed instructions in confluenca and got a simple running fs server with mod_verto which is capable of some basic webrtc. Really nice. Now I want to join this room (which is the basic 3500 room with the default video-mcu-profile) with some external h323 device. So I compiled ptlib and opalvoip and finally mod_opal and created a h323 listener. When I call the ip + room number with an external device (polycom, sony) via h323 I get voice only connections, log says the following: 2016-09-12 12:26:17.140849 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec H263-2000@90000 (pt=121) to an OPAL media format. 2016-09-12 12:26:17.140849 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec H263-1998@90000 (pt=115) to an OPAL media format. 2016-09-12 12:26:17.140849 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec VP8@90000 (pt=99) to an OPAL media format. 2016-09-12 12:26:17.140849 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec VP9@90000 (pt=99) to an OPAL media format. 2016-09-12 12:26:17.160836 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec H263@90000 (pt=34) to an OPAL media format. 2016-09-12 12:26:17.160836 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec LPC@8000 (pt=7) to an OPAL media format. 2016-09-12 12:26:17.160836 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec PROXY-VID@90000 (pt=31) to an OPAL media format. 2016-09-12 12:26:17.160836 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec H261@90000 (pt=31) to an OPAL media format. 2016-09-12 12:26:17.160836 [WARNING] mod_opal.cpp:750 {Opal Answer:12170,0000000000001} mod_opal Could not match FS codec H264@90000 (pt=97) to an OPAL media format. So I thought I should have a look about the compiled codecs. I activated mod_opal and mod_h26x in modules.conf but no succes. Log always says: 2016-09-12 12:26:17.160836 [INFO] h323.cxx:3996 {Opal Answer:12170,0000000000001} H323 SetLocalCapabilities: GSM-AMR,G.723.1,G.729,GSM-06.10,G.711-uLaw-64k,G.711-ALaw-64k,T.38,UserInput/hookflash,UserInput/basicString,UserInput/dtmf I've tried to change profile or the conference / dialplan, event set "absolute_codec_string" in dialplan/default.xml but I think I am missing something. Could anyone point me in the right direction where I can "enable" h263++ or h264 video capabilitys for an external h323 device / call via mod_opal? System is debian 8, FS and ptlib / mod_opal are checked out from git / cvs. Best regards, Juergen

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Re: [Freeswitch-users] SRTP support
by Tom Chen 04 Nov '16

04 Nov '16

Hi Brian, Thank you! Should I just add this line to vars.xml to "Accept/Offer SAVP/AVP with SAVP preferred"? <X-PRE-PROCESS cmd="set" data="rtp_secure_media=optional"/> Tom >Read the very long description on SRTP in the vars.xml file, It has some >extensive documentation on this topic. (Hint: were not a proxy.) On Wed, Oct 26, 2016 at 11:42 AM, Tom Chen <chentom60 at hotmail.com<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote: > Hello, > > > By default, Freeswitch seems always translates incoming SRTP packets to > outbound RTP only packets as the INVITE SIP message to Receiver from FS > contains only RTP/AVP audio media in SDP section even if the Caller's > INVITE has RTP/SAVP audio media only. > > > Caller to FS > > Session Initiation Protocol (INVITE) > > . . . . > > Session Description Protocol > > Session Description Protocol Version (v): 0 > > Owner/Creator, Session Id (o): 1001 625 1382 IN IP4 10.113.49.31 > > Session Attribute (a): rtcp-xr:rcvr-rtt=all:10000 > stat-summary=loss,dup,jitt,TTL voip-metrics > > Media Description, name and address (m): audio 7078 *RTP/SAVP* 96 97 98 > 99 0 8 101 100 102 > > > FS to Receiver > > Session Description Protocol > > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): FreeSWITCH 1477472443 1477472444 IN IP4 > 172.17.10.23 > > Media Description, name and address (m): audio 25206 *RTP/AVP* 0 8 101 > 13 > > > Is there a way to let FS send SRTP packets to receiver as the receiver can > accept SRTP media only? > > > Tom > > > ______

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JSAPI in Verto
by Donny Hardyanto 04 Nov '16

04 Nov '16

Hi all, Is it possible to run JSAPI via verto? Is it as powerfull as Event Socket? Is there any way to secure it? I like to deploy the verto webrtc to website, but dont want any one can abuse and control my FS box via JSAPi/verto. Thanks, Donny

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Windows build v1.6
by Gregor Nanger 03 Nov '16

03 Nov '16

Need some help. Pulled latest -v16 version and on windows build I get error: Error C2220 warning treated as error - no 'object' file generated FreeSwitchCoreLib D:\freeswitch\src\switch_rtp.c 38 It is in line 38 of file switch_rtp.c Any hints?

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ESL with PHP - last event is not received
by Melanie Treitinger 03 Nov '16

03 Nov '16

Hello, I'm using a php event socket listener based on this example (wich is linked in the freeswitch confluence): https://bitbucket.org/lrichard/freeswitcheventsocketlistener It works well so far - but the last event of an event chaine is always missing. It is only transfered when the next event takes place. For example when a conference ends we have the following events: floor-change play-file-done del-member conference-destroy But conference-destroy is not received - only when there is a next event! Is this a bug? Is the example missing something? Thanks a lot! Melanie

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503- service unavailable
by devang nathwani 02 Nov '16

02 Nov '16

Hello, Freeswitch version: 1.6.11 Number of registration on my switch: 270+ Ram: 4 GB, 1 GB is free; 64 bit debian 8 I am facing this issue, where some time i am getting continuous INVITE registration requests but freeswitch not responding to it and sometime i am getting 503 service unavailable I have gone though https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-6589/FS-6589.h… as well but not finding much helpful. here is the sip log for the same, if you want to go through. http://pastebin.com/PXz277XD Is RAM upgrade a solution? or any other? Thanks, Devang Nathwani

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New Chrome Extension for mod_verto
by Vitaly Kovalyshyn 31 Oct '16

31 Oct '16

Hi there! We've made the Chrome Extension & Application for FreeSWITCH mod-verto. It supports: Audio & Video calls via mod_verto Screen sharing Click-to-call in any web-page Address book Multiline Be free to download and test it: https://chrome.google.com/webstore/search/webitel <https://chrome.google.com/webstore/search/webitel> Best regards, Vitaly Kovalyshyn http://kovalyshyn.pp.ua/ Twitter: @kovalyshyn http://вебітел.укр/

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