July 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS6947 - Tuning Opus bandwidth
by Emrah 26 Oct '16

26 Oct '16

Hi list, I re-read FS6947 and don't understand how this problematic was addressed and the issue fixed. The scope is simple. There should be a setting in the dialplan that allows downsampling of Opus for applications that do not require the 48khz / 2 channels framework. I.e.: terminating to the PSTN with Opus to take advantage of low bandwidth and great PLC. There seems to be a lot of confusion around bandwidth in general there. It doesn't matter if the internal clock of the device is always sampling at 48khz / 2ch. There are settings that can facilitate a lower bandwidth consumption for particular use cases, and it seems the reason it is not being implemented in FS is just a matter of being confused about the intent of the 48khz 2ch base. Please revisit this issue. FS should allow tuning of Opus audio / network bandwidth in the dialplan. It would optimize greatly lots of use cases. If I'm calling the PSTN, I'd rather have my client downsample and stream at a lower bandwidth, even if my audio capture would still be at 48khz / 2ch as per the RFC, and save on bandwidth, than transcode the full 48khz spectrum into PCM on my FS and minimize processing power on the client's side. Jira here: https://freeswitch.org/jira/browse/FS-6947 Emrah

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Capturing the Current Call Audio
by Leonardo Lima Ribeiro 13 Oct '16

13 Oct '16

Hi! I’m tying to capture the current call audio but I have no idea on how to do that. My main achievement is to get the current call audio and them stream this audio bytes to an API via HTTP Post. I was doing some tries with mod_shout and mod_vlc, and I was successful in the streaming part - I tested some wavs and mp3 files. But the mainly part is that I need the current call audio, and it has no file in the system to stream right? I was thinking if there’s a command that I can get the audio bytes from the current call but I really don’t know, I could not find any option like that. Could you please help me? Thank you, Leonardo Ribeiro

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Kernel Hyperthreading ON or OFF?
by Raimund Sacherer 06 Oct '16

06 Oct '16

Hi, we have a new Freeswitch cluster with Dell R410. It has 2 CPUs (Xeon E5620, QuadCore. it's 8 cores withouth HT and 16 cores with HT. I googled with hyperthreading and freeswitch but some say turn it on it will give you a boost, others say turn it off, you will gain performance. Does anyone know if it's better to leave it on or turn it off? And some small explanation? I would be very gratefull for your help, best regards Raimund --

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Audio Control on mp3/wav
by Jai Rangi 03 Oct '16

03 Oct '16

Checking to see if it is possible to have controls on audio files played with playback or filestream. Expected controls are keys to rewind, forward, fast-forward, pause, play. Jai

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Conference recording in stereo
by Oleg Stolyar 09 Sep '16

09 Sep '16

Hi guys, is there a way to record conferences to wav files in stereo? Documentation for RECORD STEREO variable mentions Leg A and Leg B, so I am not sure it will work for a conference.

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freeswitch 1.6 without video support
by Grant Bagdasarian 19 Aug '16

19 Aug '16

Hello, Is it possible to compile and install fs 1.6 without support for video? Or is the 1.4 version the same as 1.6 with bug fixes and security updates? Regards, Grant

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Chrome v52
by Mirko Brankovic 15 Aug '16

15 Aug '16

Hi, Has anyone encountered this new problem with new version of Chrome. Maybe it is a question for the Chrome dev but just to ask :) Chrome v52 gives me bad AV handshake on FS: 2016-07-27 08:52:20.952477 [ERR] switch_rtp.c:3112 video Handshake failure 1 2016-07-27 08:52:20.952477 [INFO] switch_rtp.c:3113 Changing video DTLS state from HANDSHAKE to FAIL 2016-07-27 08:52:20.982513 [ERR] switch_rtp.c:3112 audio Handshake failure 1 2016-07-27 08:52:20.982513 [INFO] switch_rtp.c:3113 Changing audio DTLS state from HANDSHAKE to FAIL On v51 fingerprint is verified: 2016-07-27 09:03:02.072474 [INFO] switch_rtp.c:3119 Changing video DTLS state from HANDSHAKE to SETUP 2016-07-27 09:03:02.122569 [INFO] switch_rtp.c:3028 video Fingerprint Verified. 2016-07-27 09:03:02.122569 [INFO] switch_rtp.c:3644 Activating Video Secure RTP SEND -- Regards, Mirko

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Fwd: Freeswitch call via WebRTC not work with Chrome 52
by Hai Bui Duc Ha 12 Aug '16

12 Aug '16

Hi all, - I have problem with Freeswitch call via WebRTC on Chrome 52, I can't establish the call and server appear log: *2016-07-31 17:28:01.977603 [ERR] switch_rtp.c:2917 audio Handshake failure 1* - I have already researched on Internet: https://groups.google.com/forum/?fromgroups#!topic/bigbluebutton-dev/5lR5WW… - They told that: I need to update OpenSSL version to 1.0.2 or later (currently is 1.0.1e-2+deb7u21) - Because, my OS(Debian Wheezy 7.11) don't have OpenSSL 1.0.2 on the repositories, so that I built this lib from source openssl-1.0.2h and also built libcurl 7.42. - After that, I re-built freeswitch like the guide at link https://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#FreeSWITCH… - I use FreeSWITCH Version 1.4.26+git~20151119T204804Z~66cd7d6b1a~64bit - But now, I still not work. You have any idea ? Thank for your support! Best regards, Hai Bui -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876

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httapi - dial/bridge only working when it comes first
by Don Hawkins 03 Aug '16

03 Aug '16

I'm using httapi and everything is working well for the most part. The one thing I noticed is that whenever I do anything before trying to "Dial" or bridge a call to the gateway: 1. The first thing never happens, even though the console says it did. 2. The outbound call "Dial" won't work, it hangs for about 60 seconds and then returns a NORMAL_TEMPORARY_FAILURE every time. Things to note: - If I start with the dial request (B below) then it connects the call fine, it just can't come after anything else. - If I never return a request to dial (pretend B below starts a conference instead) then everything works fine, the TTS works before and the conference works after. Here is an example: A. When FS makes it's first request to my URL I return this: <document type="text/freeswitch-httapi"> <work> <speak action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabl…" name="speak" loop="0" engine="flite" voice="slt" text="We will now begin to dial a number. This is just a test." language="en"> </speak> </work> </document> B: When FS makes it's next request to my action URL (same URL with different parameters) I return the dial command and attempt to connect to the outbound gateway. <document type="text/freeswitch-httapi"> <work>  <execute application="set" data="ringback=http://blabla.com/beat.wav "></execute>  <execute application="set" data="effective_caller_id_name=My Name"></execute> <execute application="set" data="effective_caller_id_number=12144635555"></execute>  <execute application="export" data="nolocal:codec_string=G711"></execute> <execute action=" http://blabla.com/API/callflow/?customer_next_action_url=http%3A%2F%2Fblabl… " application="bridge" data="sofia/gatewayAlcazar/12144635555"> <execute application="answer"/> </execute> </work> </document> What happens? It never speaks the text (although the console says it did) and when dialing out I get "NORMAL_TEMPORARY_FAILURE" and the call never connects. Keep in mind, if the second thing I return (B above) is NOT dial then everything works fine, it will speak the text, play a file, start a conference, or any other application. This is the weirdest thing to me and of course almost everything is useless if you can't connect a call at some point, if anyone has any advice I'd be highly appreciative.

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Determine When Bridge Attempt is Complete
by Colin Morelli 03 Aug '16

03 Aug '16

Hey, I'm looking to determine when a bridge request is complete. I'm using mod_amqp to receive events from Freeswitch and using that data to update a separate database. The problem is that, so far, I've only been able to determine when a bridge attempt fails in a channel_execute_complete event. I'm really trying to avoid sending that event over mod_amqp because there are *a ton* of those events generated for every call. Is there any other way to determine when a bridge attempt fails? We've got CHANNEL_BRIDGE & CHANNEL_ANSWER to determine when it's successful - would be great if anyone knows for the other case. Best, Colin

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