May 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue
by sachin 08 May '17

08 May '17

Hello Gregor,I have upgraded the openssl version to 1.1.0e from the following link. But the issue remains the same.http://slproweb.com/download/Win64OpenSSL-1_1_0e.exeIs the above way correct to upgrade the open ssl.Thanks and Regards,SachinFrom: Brian West <brian(a)freeswitch.org>Sent: Fri, 05 May 2017 19:26:58To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org>Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue Clearly that isn't the case, I suspect a define is missing to actually enable ECDSA at this point.  On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger <gregor(a)infomedia.si> wrote: Brian, isn't this solved in 1.6.16?   FS-10037 [core] Update OpenSSL to version 1.0.2k for Windows  2017-05-05 15:26 GMT+02:00 Brian West <brian(a)freeswitch.org>: OpenSSL on the windows build needs to be updated.  https://freeswitch.org/jira/browse/FS-9510  On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger <gregor(a)infomedia.si> wrote: This error is familiar to me, I think so, if I remembered correctly.    I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it.  2017-05-04 19:15 GMT+02:00 Anthony Minessale <anthony.minessale(a)gmail.com>: FS 1.5 sounds like a bad plan.Try latest FS 1.6 or master.    On Thu, May 4, 2017 at 12:04 PM, Ítalo Rossi <italo(a)freeswitch.org> wrote: Old openssl version maybe?Em qui, 4 de mai de 2017 às 11:19, sachin <snehsach(a)rediffmail.com> escreveu: Hello All,I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN.The clients are getting registered over wss. I have created self signed certificates.  In var.xml I have set the codecs setting as follows  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA">  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA">I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 12017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001@df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER]My setup is as followsSIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> LinphoneI am attaching the logs for the reference.fs-logleve9.txt : Debug trace with loglevel =9fs-sip-trace.txt : Sip tracePlease let me know what could the issue and pointers to resolve the same.Thanks and RegardsSD_________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org_________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org   -- Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬  ☞ http://freeswitch.org/  ☞ http://cluecon.com/  ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ 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 _________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org   -- Gregor Nanger   CTO t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si _________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org   -- Brian Westbrian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST)Allison prompts for FreeSWITCH: https://www.gofundme.com/allison-prompts-for-freeswitch Got Bugs? Report them here! | Reddit: /r/freeswitch T: 19184209001 | F: 19184209002 | M:+1918424WEST (9378)Skype:briankwest_________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org   -- Gregor Nanger   CTO t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si _________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org   -- Brian Westbrian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST)Allison prompts for FreeSWITCH: https://www.gofundme.com/allison-prompts-for-freeswitch Got Bugs? Report them here! | Reddit: /r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)Skype:briankwest_________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.comOfficial FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.comFreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org

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Freeswitch stops handling register requests
by Markus Bönke 08 May '17

08 May '17

Hello, We faced today the following situation: Freeswitch suddenly stopped to respond to answer to REGISTER requests. At the same time it created more and more channels. A lot of them „hung" in status „RINGING“, after some time all calls were rejected with 503 Maximum Calls In Progress. After a restart, everything went back to normal I cannot find any WARNING or ERRROR/CRIT messages in the log. Also there is nothing special in the syslog at the time the issue appeared. We are using freeswitch 1.6.17 on debian jessie installed from debian packages. Calls are handled via esl / mod_event socket. Is there any known issue with this version? Is there anything else I can check? Thanks and Regards Markus

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video call using H264
by Gauri Kshirsagar 08 May '17

08 May '17

Hi, I am new to freeswitch. I installed Freeswitch on CentOS 7 using rpm. I tried making video call using H264 but it failed. show codec on fs_cli does not list H264. What are the configurations required to make a video call ? Rgds, GK

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Trickle ICE using Verto
by Adam Ben-Ayoun 08 May '17

08 May '17

Hi, We are currently using SIP for WebRTC for our app (web, iOS and Android implementations). ICE gathering can sometimes be slow especially on mobile and we are looking for ways to speed this up. We thought about using trickle ICE but it's not possible/easy using SIP/Freeswitch. My question is if we can do trickle ICE using verto. Thanks, Adam

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One way audio after sometime
by Gaetano Volpe 08 May '17

08 May '17

Hi guys, I'm running the last stable version of Freeswitch in my environment. FreeSWITCH Version 1.6.14~64bit ( 64bit) I've a very strange problem with a group of extensions of sales department An isdn pri line is provided by a cisco router (2921) through a sip trunk. When the calls volume increases, suddenly some phones experience one way audio (the customer hears, sellers don't). When it happesn the only way to get the phone back to work is to reboot it. We're running all polycom vvx 300 phones. What I've tried so far: 1. Change the codec to g.711 2. Upgrade freeswitch 3. Upgrade Polycom firmware 4. Downgrade Polycom firmware I've captured packets with wireshark packets and I can see both audio channels in the call trace. So I guessed it could be the phones but it's very strange. What do you think about it? Thanks *Gaetano VolpeCEO, CTO & Systems Analyst*g.volpe(a)studiofox.it *- Mob.+39.3485260691* -- <http://www.studiofox.it> *STUDIO FOX SERVICE di G.Volpe* Via Spineto, 24 - 70038 - Terlizzi (Bari) - Italy Ph. +39.0803540043 - Fax +39.0803515004 C.F. VLPGTN83E10L109S - P.Iva IT05803550721 www.studiofox.it La presente comunicazione ed i suoi eventuali allegati potrebbero contenere informazioni legate al segreto professionale o dati soggetti alla vigente Legge sulla Privacy e sono comunque strettamente riservati e confidenziali per il destinatario. In mancanza di autorizzazione è vietata la lettura, la riproduzione, la diffusione e l'estrazione di copie. Pertanto, nel caso in cui aveste ricevuto per errore la presente comunicazione, Vi preghiamo di provvedere a distruggerla e di contattare immediatamente i nostri uffici ai numeri su indicati. Grazie. This communication and its attachments, if any, may contain information covered by professional secrecy or data subject to current privacy law, and they are intended only for the use of the individual to whom it is addressed. They may contain information that is legally privileged, confidential or exempt from disclosure. Reading, reproduction, disclosure, and extraction of copies are prohibited unless expressly authorized. If you have received this communication in error, please notify immediately our offices at the contact details specified above and delete the communication without retaining any copies.

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mod_distributor from db
by Grant Bagdasarian 08 May '17

08 May '17

Hello, Is it possible to load the gateways of a sip profile from a database and the distributor config also from a database, using built in modules/functions? Regards, Grant

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Freeswitch and IoT
by Marcel Haldemann 06 May '17

06 May '17

Hi Guys, is it possible to create a channel using event socket (without calling a number) ? LG, Marcel

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Script language choice
by Oivvio Polite 06 May '17

06 May '17

There's a page about script language choice on the wiki. There's a caveat at to top warning that the information is stale. https://freeswitch.org/confluence/display/FREESWITCH/Script+Language+Choice Anyway, it states that Lua is the preferred scripting language due efficacy. In a scenario where I'm not expecting very high loads are there other reasons for going with Lua anyway? I'm thinking that it's probably more battle tested if it's the most common choice among FS-developers. regards Oivvio

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Video pixelated problem inconferences calls.
by Andres Gomez 05 May '17

05 May '17

Hello I have a problem with videoconference and I applied the changes that are made in the mails, however I did not work, at the moment of connecting 3 or more to videoconference the image is pixelated. My network schema is as follows softphones <---- lan ----> FS <-------- MPLS 20M -------> Video Hardphones . My configuration is as follows: varls.xml  <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_in=6mb"/> <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_out=6mb"/> conference.xml.conf <profile name="video-mcu-stereo"> <param name="domain" value="$${domain}"/> <param name="rate" value="48000"/> <param name="channels" value="2"/> <param name="interval" value="10"/> <param name="energy-level" value="200"/>   <param name="video-quality" value="1"/> <param name="muted-sound" value="conference/conf-muted.wav"/> <param name="unmuted-sound" value="conference/conf-unmuted.wav"/> <param name="alone-sound" value="conference/conf-alone.wav"/> <param name="moh-sound" value="$${hold_music}"/> <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/> <param name="exit-sound" value="tone_stream://%(500,0, 300,200,100,50,25)"/> <param name="kicked-sound" value="conference/conf-kicked.wav"/> <param name="locked-sound" value="conference/conf-locked.wav"/> <param name="is-locked-sound" value="conference/conf-is-locked.wav"/> <param name="is-unlocked-sound" value="conference/conf-is- unlocked.wav"/> <param name="pin-sound" value="conference/conf-pin.wav"/> <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/> <param name="caller-id-name" value="$${outbound_caller_name}"/> <param name="caller-id-number" value="$${outbound_caller_id}"/> <param name="comfort-noise" value="false"/> <param name="conference-flags" value="video-floor-only|rfc- 4579|livearray-sync|minimize-video-encoding|video-muxing-personal-canvas"/> <param name="video-mode" value="mux"/> <param name="video-layout-name" value="3x3"/> <param name="video-layout-name" value="group:grid"/>  <param name="video-canvas-size" value="1280x720"/> <param name="video-canvas-bgcolor" value="#333333"/> <param name="video-layout-bgcolor" value="#000000"/> <param name="video-codec-bandwidth" value="4mb"/> <param name="video-fps" value="30"/> </profile> Regards

1 0

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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