April 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Video frame capture
by William Suffill 01 Apr '15

01 Apr '15

ffmpeg seems to be a common option to export frames as images from FLV among other formats. That might work in your case here as well Something like http://stackoverflow.com/questions/3827611/ffmpeg-to-capture-screenshot-fro… On Wed, Apr 1, 2015 at 5:40 AM, Marino Mileti <marino.mileti(a)alice.it> wrote: > Hi guys, > > > > i'm trying to implement something like "preview" feature on video mailbox; > just a simple picture of a frame of .fsv file. > > Unfortunately i've no idea regarding .fsv file J > > > > Is it possible to "extract" an image from .fsv file? > > > > Any type of help is very appreciated J > > > > Many thanks > > *Marino Maria Mileti* > > *marino.mileti(a)alice.it <marino.mileti(a)alice.it>* > > > > *[image: cid:006a01cb6b0e$67eecdae$_CDOSYS2.0]**Reduce your energy > consumption and keep polar bears on ice!* > > > > > ------------------------------ > <http://www.avast.com/> > > Questa e-mail è priva di virus e malware perché è attiva la protezione avast! > Antivirus <http://www.avast.com/> . > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

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Video frame capture
by Marino Mileti 01 Apr '15

01 Apr '15

Hi guys, i'm trying to implement something like "preview" feature on video mailbox; just a simple picture of a frame of .fsv file. Unfortunately i've no idea regarding .fsv file J Is it possible to "extract" an image from .fsv file? Any type of help is very appreciated J Many thanks Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è priva di virus e malware perché è attiva la protezione avast! Antivirus. http://www.avast.com

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Caller ID prefix missing
by Paul Atreides 01 Apr '15

01 Apr '15

Hi, when I place a call over the gateway to the PTSN I see the dialed number without the leading 0 prefix in the history of my Grandstream GXP2140. Is there a way to add the 0 back again? Thank you <include> <extension name="isdn-outbound"> <condition field="destination_number" expression="^0(\d+)$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/isdn/$1"/> </condition> </extension> </include>

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OPUS at 48 khz vs OPUS at 8 khz
by doancea 01 Apr '15

01 Apr '15

Hi All, We've noticed a high load on our transcoding servers when doing OPUS<->PCMA transcoding. All our client software is using opus@48khz, but inside the audio payload we have narrowband audio (forced with OPUS_SET_BANDWIDTH(OPUS_BANDWIDTH_NARROWBAND) ) . Questions : 1. How can we avoid resampling ? We've noticed this log line showing twice for each call: 29128b92-d842-11e4-877f-9bd23fac7445 2015-04-01 07:39:14.798759 [NOTICE] switch_core_io.c:1287 Activating write resampler If we modify the client software to use opus@8khz and leave freeswitch as it we have very bad audio (nothing intelligible). But If we modify the client software to use opus@8khz and change mod_opus.c in freeswitch so it would use only OPUS@8khz then we have proper audio. 2. We have noticed that there is a "samplerate" param in fmtp (which is not in this RFC: https://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03 ) it's used only for configuring the decoder. Why only for the decoder ? If "samplerate" is set then it's passed to opus_encoder_create() , but not to opus_decoder_create(). 3. How about using sprop-maxcapturerate to set the sample rate of the decoder ? Regards, Dragos Oancea

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Announcement: Kamailio becomes systemd-rtc-server
by Alex Balashov 01 Apr '15

01 Apr '15

For immediate release: ATLANTA, GA (1 April 2015)--Evariste Systems LLC, an Atlanta-based software vendor specialising in Kamailio-based service delivery solutions for the VoIP ITSP market, is pleased to announce that it, in collaboration with Red Hat Software and Ringfree Communications, has finalised the absorption of the Kamailio SIP Server into the 'systemd' system management platform for Linux. The new component shall be called 'systemd-rtc-server', or 'Systemd Real-Time Communication Server'. Alex Balashov, principal of Evariste and leader of the tri-vendor collaboration effort, will officially announce the handover of the reigns of the Kamailio project to the personal leadership of Lennart Poettering at the upcoming Systemd Real Time Communications World conference, to be held in Berlin on 27-29 May of this year. John Knight, Director of GNOME 3 Integration and part-time usability consultant at Ringfree Communications, based in Hendersonville, North Carolina, summarised the triumphs of the long-standing integration effort. Remarked Knight: "The industry has recognised for years that a SIP proxy is a basic building block in the 'init' subsystem of any Linux host. In this age of multimedia communication with voice and video, it was a travesty that systemd handled time synchronisation, network configuration, login management, logging, and console, but not SIP message routing." Sean McCord, a veteran partner at Atlanta-based integrator CyCORE & Docker, was quick to concur: "SIP calls are much easier to troubleshoot with binary logs. Combined with packet captures of TLS-encrypted WebRTC calls, systemd-journald is the ultimate call setup troubleshooting methodology of the responsive, kinetic enterprise." To support the integration of Kamailio into the ecosystem of every major Linux distribution, Evariste has released new 'dbus_api' and 'pulseaudio' modules for the project. Balashov stated, "We fully expect to use the D-Bus API to achieve gnome-session integration with systemd-rtc-server-usrloc, but we aren't going to leave Windows users behind; KamailioSvcHost.exe will support Domain Controller policies for G.722 in Active Directory forests." Despite an aggressive delivery timeline by the tri-vendor consortium behind systemd-rtc-server, industry commentators have widely lambasted the fact that it took so long for Kamailio to become integrated into systemd. Fred Posner, solutions architect at The Palner Group in Fort Lauderdale, Florida, recently wrote in a widely-publicised blog post: "sr-dev have been keeping their heads in the sand for too long. For years now, it has been completely obvious and self-evident to anyone with half a brain that all kinds of VoIP software should be included in systemd. It's a basic building block of the whole OS, having absorbed functionality previously provided by all kinds of packages like util-linux and wireless-tools." John Knight of Ringfree accepted the criticism readily, but advocated a forward-thinking orientation focused on breaking with the uncertainty of the past: "In the absence of a SIP component for routing calls to the PSTN, some people thought, 'systemd has no clear direction apart from the whims of its developers, and is a perpetually moving goal post.' Well, a SIP server should put an end to that whole discussion; that's exactly what was missing, and now that we have systemd-rtc-server, we've eliminated all doubts about the coherence, conceptual integrity and finality of systemd." -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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