July 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Can't load mod_lua.so on Freeswitch v1.6.20
by louie liu 30 Jul '18

30 Jul '18

Hi , I have installed Freeswitch 1.6.20 from RPM package, I've setup lua.conf.xml to run a lua script "gen_dir_user_xml_dbh.lua" to load user directory information from mysql when FS starts up, however the FS logs indicates it couldn't find mod_lua.so share library file. Since Centos 7.5 OS came with lua 5.1.4 preinstalled. If FS move to use the system lua, why does it still reference the mod_lua.so. How to make it work under current FS? 2018-07-27 20:29:52.476412 [CRIT] switch_loadable_module.c:1522 Error Loading module /usr/lib64/freeswitch/mod/mod_lua.so **libssl.so.6: cannot open shared object file: No such file or directory** lua -v Lua 5.1.4 Copyright (C) 1994-2008 Lua.org, PUC-Rio Lua.conf.xml <configuration name="lua.conf" description="LUA Configuration"> <settings>       <param name="xml-handler-script" value="gen_dir_user_xml_dbh.lua"/> <param name="xml-handler-bindings" value="directory"/> </settings> </configuration>

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Re: [Freeswitch-users] [OpenSIPS-Users] FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and Conference Service, ClueCon Presentation Slides
by Bogdan-Andrei Iancu 26 Jul '18

26 Jul '18

Hey, that's a good presentation Giovanni, thanks for sharing it with us !! Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Bootcamp 2018 http://opensips.org/training/OpenSIPS_Bootcamp_2018/ On 07/25/2018 07:29 PM, Giovanni Maruzzelli wrote: > Hello fellow RTCers, > > just finished presenting about scaling videoconferencing, chat and > moderation/direction controls in SIP, with web clients, deskphones, > smartphone apps, server push (google fcm) via cordova and SIP.js at > www.cluecon.com <http://www.cluecon.com>. > > Here's the slides of my presentation: > > http://178.79.181.125/Maruzzelli_FreeSWITCH_OpenSIPS_WebRTC_Pure_SIP_Video_… > > or, shorter, > > http://178.79.181.125/Maruzzelli_ClueCon_2018.pdf > > > From ClueCon blurb: > > > FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and > Conference Service > > On top of traditional SIP deskphones and softphones, WebRTC let us > extend the reach of SIP to all browsers, and to smartphone apps that > wake up with a server push. OpenSIPS has superb SIP and WebRTC support > and can protect, balance and scale FreeSWITCH insuperable > conferencing, video MCU, media switching/mixing, SIP SIMPLE message > processing capabilities. We will see how to build and implement such a > complete platform, touching both on servers and on clients. Hint: > SIMPLE, the SIP Instant Messaging Protocol, can be leveraged to > transport statuses, commands and chats between participants and > administrators of SIP audio/video calls/conferences. > > > Giovanni Maruzzelli > > > Principal at OpenTelecom.IT > > ------------------------------------------------------------------------ > > Giovanni is a consultant for the Telco sector, developing software and > training courses for FreeSWITCH, SIP, WebRTC and Kamailio. He is > heavily engaged with FreeSWITCH, of which he wrote the interface with > Skype and with cellular phones. An Internet tech pioneer, in 1996 > Giovanni was cofounder of Italia Online, the most popular Italian > portal and consumer ISP, and architect of its Internet technologies – > www.italiaonline.it <http://www.italiaonline.it> Then supervisor of > Internet operations and architect of the first engine for paid access > to www.ilsole24ore.com <http://www.ilsole24ore.com> , the most read > financial newspaper in Italy and to its databases (migrated from > mainframe). After that, he was CEO of venture capital funded Matrice, > developing Telemail unified messaging and multi language phone access > to email (Text To Speech), and CTO of incubator funded Open4, an open > source managed applications provider. Then he was for two years in > Serbia as Internet and Telecommunication Investment Expert for World > Bank – IFC. Since 2005 he’s based in Italy, and serves ICT and Telco > companies worldwide. > > > -giovanni > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > > > _______________________________________________ > Users mailing list > Users(a)lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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(no subject)
by Mi Ke 26 Jul '18

26 Jul '18

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ZRTP trunk
by Mi Ke 26 Jul '18

26 Jul '18

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FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and Conference Service, ClueCon Presentation Slides
by Giovanni Maruzzelli 25 Jul '18

25 Jul '18

Hello fellow RTCers, just finished presenting about scaling videoconferencing, chat and moderation/direction controls in SIP, with web clients, deskphones, smartphone apps, server push (google fcm) via cordova and SIP.js at www.cluecon.com. Here's the slides of my presentation: http://178.79.181.125/Maruzzelli_FreeSWITCH_OpenSIPS_WebRTC_Pure_SIP_Video_… or, shorter, http://178.79.181.125/Maruzzelli_ClueCon_2018.pdf >From ClueCon blurb: FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and Conference Service On top of traditional SIP deskphones and softphones, WebRTC let us extend the reach of SIP to all browsers, and to smartphone apps that wake up with a server push. OpenSIPS has superb SIP and WebRTC support and can protect, balance and scale FreeSWITCH insuperable conferencing, video MCU, media switching/mixing, SIP SIMPLE message processing capabilities. We will see how to build and implement such a complete platform, touching both on servers and on clients. Hint: SIMPLE, the SIP Instant Messaging Protocol, can be leveraged to transport statuses, commands and chats between participants and administrators of SIP audio/video calls/conferences. Giovanni MaruzzelliPrincipal at OpenTelecom.IT ------------------------------ Giovanni is a consultant for the Telco sector, developing software and training courses for FreeSWITCH, SIP, WebRTC and Kamailio. He is heavily engaged with FreeSWITCH, of which he wrote the interface with Skype and with cellular phones. An Internet tech pioneer, in 1996 Giovanni was cofounder of Italia Online, the most popular Italian portal and consumer ISP, and architect of its Internet technologies – www.italiaonline.it Then supervisor of Internet operations and architect of the first engine for paid access to www.ilsole24ore.com , the most read financial newspaper in Italy and to its databases (migrated from mainframe). After that, he was CEO of venture capital funded Matrice, developing Telemail unified messaging and multi language phone access to email (Text To Speech), and CTO of incubator funded Open4, an open source managed applications provider. Then he was for two years in Serbia as Internet and Telecommunication Investment Expert for World Bank – IFC. Since 2005 he’s based in Italy, and serves ICT and Telco companies worldwide. -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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video on conference with PC and MAC browser works but fail with iPhone safari via webrtc
by 薛光宏 25 Jul '18

25 Jul '18

Hi All, here is our simple architecture <client A>---<FreeSWITCH> <client B>----/ - client A and B are both registered on FreeSWITCH via websocket (using sipjs 0.9.2) - we enable video-mode= transcode and load mod_av on FreeSWITCH. - if client A uses chrome on PC and client B use safari on *MAC*, they can see each other, talk to each other in the same conference. - if client A uses chrome on PC and client B use safari on *iPhone*(8plus, iOS 11.4), they can NOT see each other, talk to each other in the same conference. I though safari on MAC works, then safari on iPhone should work, but not. Is it possible to fix that with changing configuration of FreeSWITCH? or any suggestion? Thanks in advance. FreeSWITCH Version 1.6.20 Ubuntu Server Ubuntu 14.04.1 LTS 64bit libav-11.4 or libav-12 See below configurations: 1. enable video-mode= transcode <param name="conference-flags" value="transcode-video|minimize-video-encoding"/> <param name="video-mode" value="transcode"/> 2.enable VP8/H264 codec support <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,VP8,H264"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,VP8,H264"/> 3.enable mod_av and disable mod_h26x <load module="mod_av"/>

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segmentation fault when rescanning profile
by Juan Pablo L 24 Jul '18

24 Jul '18

Hello, i m getting a segmentation fault everytime i try to do anything with a sofia profile, for example everytime i try to rescan or start a profile. Analysing the core dumped the problem seems to be in "switch_ci_hashfunc_default": Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `freeswitch'. Program terminated with signal SIGSEGV, Segmentation fault. #0 switch_ci_hashfunc_default (char_key=0x0, klen=0x7f1d309313e8) at src/switch_apr.c:112 112 for (p = key; *p; p++) { (gdb) bt #0 switch_ci_hashfunc_default (char_key=0x0, klen=0x7f1d309313e8) at src/switch_apr.c:112 #1 0x00007f1d30a28013 in xml_directory_db_search (section=0x0, tag_name=0x7f1d309313e8 "\377\377\377\377\377\377\377\377", key_name=0x0, key_value=0x0, event_params=0x7f1d0c16de00, user_data=0x0) at xml_directory.c:155 #2 0x00007f1d71e8f0db in switch_xml_locate (section=0x7f1d69468101 "directory", tag_name=0x0, key_name=0x0, key_value=0x0, root=0x7f1d0c16de00, node=0x0, params=0x7f1d0c16de00, clone=SWITCH_FALSE) at src/switch_xml.c:1675 #3 0x00007f1d69351ce3 in config_sofia (reload=(unknown: 201695600), profile_name=0x7f1d309313e8 "\377\377\377\377\377\377\377\377") at sofia.c:5890 #4 0x00007f1d69336d8c in cmd_profile (argv=0x7f1d309318d8, argc=2, stream=0x7f1d30933db0) at mod_sofia.c:3324 #5 0x00007f1d69336445 in sofia_function (cmd=<optimized out>, session=<optimized out>, stream=0x7f1d30933db0) at mod_sofia.c:4327 #6 0x00007f1d71e2628e in switch_api_execute (cmd=<optimized out>, arg=<optimized out>, session=0x0, stream=0x7f1d30933db0) at src/switch_loadable_module.c:2567 #7 0x00007f1d71dbb9e0 in switch_console_execute (xcmd=0x7f1d0c009f20 "sofia", rec=0, istream=0x7f1d30933db0) at src/switch_console.c:395 #8 0x00007f1d71dbbb51 in switch_console_process (xcmd=0x7f1d0c001350 "sofia profile internalgw rescan") at src/switch_console.c:323 #9 0x00007f1d71dbbc68 in console_thread (thread=0x0, obj=0x7f1d309313e8) at src/switch_console.c:1079 #10 0x00007f1d720be010 in dummy_worker (opaque=0x194a878) at threadproc/unix/thread.c:151 #11 0x00007f1d70c8d064 in start_thread (arg=0x7f1d30934700) at pthread_create.c:309 #12 0x00007f1d709c262d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:111 i m using : Linux sbc02 3.16.0-6-amd64 #1 SMP Debian 3.16.57-2 (2018-07-14) x86_64 GNU/Linux FreeSWITCH Version 1.6.20+git~20180507T185611Z~43a9feb7f8~64bit (git 43a9feb 2018-05-07 18:56:11Z 64bit) there is absolutely no warnings or errors when Freeswitch is booting up ... thank you for your help!!

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BYE not sent in case of RTP media inactivity with proxy_media= true in freeswitch
by vivek＠advaitamtech.com 22 Jul '18

22 Jul '18

Hi, We are running freeswitch as proxy_media = true (proxy mode). In case of RTP media inactivity, Freeswitch is not sending BYE. In sofia profile "rtp-timeout-sec" is enabled. Still freeswitch is not sending BYE in case of media inactivity. Anyone faced this issue?. Please help. Thanks, Vivek.

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mod_cdr_mysql
by Francesco Facco de Lagarda 22 Jul '18

22 Jul '18

I installed my fs by apt-get on Debian jessie Is it possible to et mod_cdr_mysql from apt or any other way? Otherwise can I compile freeswitch from source over my existing installation without breaking everything?

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Influencing "Update Callee ID" behavior
by Thilo-Alexander Ginkel 19 Jul '18

19 Jul '18

Hello everyone, when placing an external outbound call via mod_sofia I am seeing the callee's number and name being updated as soon as the called party picks up the call: 2017-12-08 10:19:24.933767 [INFO] sofia.c:1279 sofia/external/040xxx Update Callee ID to "Outbound Call" <40xxx> This seems to override what has been previously set as sip_callee_id_name (e.g., using <action application="set" data="sip_callee_id_name=${db(select/cid/${2})}"/>). There are two problems that I'd like to solve: 1. When sip_callee_id_name was previously set (e.g., via DB lookup) I'd like to keep that value (or repeat the lookup) so that the callee's name remains intact. 2. Certain destinations incorrectly strip the loeading zero from the callee's number, which I'd like to avoid displaying as is. Is there any way to influence what is displayed as name and number, i.e. keep the previously set sip_callee_id_name and apply some kind of normalization to the reported number to cope with the missing leading zero? Thanks, Thilo

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