September 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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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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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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Dangerous prototype question for Seven Du
by Christoph Russow 24 Oct '16

24 Oct '16

Hi, i've got a question for Seven Du. I was very impressed by your dangerous prototype on ClueCon 2016 and i'm wondering if you would share some more details about it. For example some background about the UI and how you executed the generated code within Freeswitch. Best Regards Christoph Russow -- ________________________________________________________________ EMTEX GmbH Christoph Russow Software Engineer Bischof-Otto-Weg 9 D-91086 Aurachtal Geschäftsführer: Markus Enzinger Tel. +49 9132 7490 0 Sitz der Gesellschaft: 91086 Aurachtal Fax. +49 9132 7490 900 Amtsgericht Fürth: HRB6804 ________________________________________________________________

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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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Opensips integration questions
by Schneur Rosenberg 12 Oct '16

12 Oct '16

Hi, I'm pretty new to Freeswitch, I built a complete solution using OpenSIPS/Asterisk and I'm in the process of replacing Asterisk with Freeswitch. My current system is built that OpenSIPS act as a registrar and as a load balancer, the rest is done on a cluster of Asterisk machines, which as I said before is being rewritten to Freeswitch, in Freeswitch I'm using xml_curl for my dialplan and call routing, I'm using Intralanman project to handle the sip users. With Asterisk I had OpensSIPS as a trunk and I know that everything coming from that trunk, is either a DID or a authenticated outgoing call, Asterisk and OpenSIPS shared the user table, Asterisk recognized the sip user name from the FROM field and used all the variables etc from its own sip user table as if the user has registered directly to itself, I had some issues with blind transfers but I was able to do some manipulations and DB queries to handle it fine. I tried doing the same with Freeswitch, I let the ACL allow the ip from Freeswitch, but Freeswitch would not use the variables from the user like Asterisk did, for example accountcode, user_context etc, I was able to partially fix that by having my web server do a db query and then using the "set" application set all the variables, but I'm not sure if its the proper solution, does anyone have a better solution? will setting my OpenSIPS as a gateway instead of merely allowing it in with ACL work better? right now I'm using the application "execute_extension" to send it to the proper context that I've pulled from the DB, I dont like the hack and if there is a better solution I would like to know. One more major problem that I have is NAT related, when I place a outgoing call for example from a phone which is behind a NAT (all of our customers phones are behind NAT) the call connects and audio works 2 ways, but when the called party hangs up, the carrier sends the BYE to Freeswitch, freeswitch then sends it to my OpenSIPS which is supposed to send it to my clients phone, but Freeswich writes the RURI with the natted address, the RURI is BYE sip:101@192.168.0.11:5062 SIP/2.0. and my OpenSIPS does not know what to do with it and ignores it, my Asterisk replaced the RURI with the received address in the VIA, so same RURI would be BYE sip:101@64.69.47.175:5062 SIP/2.0. which worked fine, same problem happens with all subsequent requests if user does not hang up, I need a solution for this major issue. thank you S. Rosenberg

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Using presence or state to avoid intercom barge-in
by Phil Quesinberry 09 Oct '16

09 Oct '16

Hi, We're running into an issue with Cisco SPA50x phones where an intercom call to an extension either places the currently active call on hold or hangs up on it and switches to the intercom call. We'd like to not have this happen if the destination extension is in use. The only way I know of to do this would involve reconfiguring all of the phones to share presence and then check it in the dialplan before bridging. Is there a convenient "no-barge" variable or similar that can be set so FS won't send the invite if the extension is in use or a way to check the extension's state from the dialplan? I'm guessing that's a NO but I had to ask! Alternatively, is there a way to get the Cisco phones not to interrupt the call in progress when a call comes in? Instead of setting sip_auto_answer=true, I've also tried setting sip_h_Call-Info=<sip:ip.address.of.server>;answer-after=0 which appears to stop this behavior on the Aastra phones but not the Cisco phones, although I'm not entirely sure about this as I now can't seem to reproduce the undesired behavior on the Aastra phones... Dialplan entry for the intercom as it stands is as follows: <extension name="extension-intercom-fix" continue="true"> <condition field="destination_number" expression="^\*8(\d{2,7})$" > <action application="set" data="dialed_extension=$1" /> <action application="bridge" data="user/${dialed_extension}@${domain_name}" /> <action application="export" data="sip_auto_answer=true" /> </condition> </extension> Any thoughts or advice would be appreciated. Thanks! Phil Quesinberry Q Systems Engineering, Inc. Embedded Systems, Telecom, IT (410) 969-8002 http://www.qsystemsengineering.com <http://www.qsystemsengineering.com/>

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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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