August 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Change SIP Contact header on b-leg
by Кирилл Преслер 22 Feb '19

22 Feb '19

Provider not accepting calls because of "bad" Contact header sip:gw+ss@10.50.244.15:5060;transport=udp;gw=ss ITSP needs sip:username@10.50.244.15:5060, where username should be passed from dialplan for every call and there is no params. I've tried following variables with set and export, but nothing seems to work. sip_contact_user sip_outgoing_contact_uri sip_invite_contact_params

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Re: [Freeswitch-users] stream a file multicast with mod_esf
by Sluschny, Thomas 10 May '18

10 May '18

Hi Brian, i thought if i can stream from portaudio it is almost the same with streaming from file, so it should working already now. Is this not the design idea of channels and media to do so? regards, thomas PS: sry for improper formatted mail, i cant reply at the moment and have to copy mail from archive :( ________________________________ >From brian at freeswitch.org Thu Feb 12 07:07:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:07:36 -0600 Subject: [Freeswitch-users] stream a file multicast with mod_esf Message-ID: <6F68FEDB-8976-44D4-9A67-F871EC7D5120(a)freeswitch.org <mailto:6F68FEDB-8976-44D4-9A67-F871EC7D5120@freeswitch.org> > esf is for multi cast paging... it currently won't let you play files... we would have to create a multicast playback application. /b On Feb 12, 2009, at 8:00 AM, Sluschny, Thomas wrote: > Hi, > > i want to stream a file per IP multicast with mod_esf. > > I can stream IP multicast with: > pa call stream XML > and in XML dialplan: > <extension name="stream"> > <condition field="destination_number" expression="^stream$"> > <action application="answer"/> > <action application="esf_page_group" data="224.168.168.168 34567"/> > <action application="esf_page_group" data=""/> > </condition> > </extension> > and i can also play files with 'playback' app, > > BUT: how can put these 2 things together? > > May be its trivial, but i cant get it make working with 'originate' > or 'uuid_broadcast' or 'bridge', 'transfer' and so on ... > > Thanks in advance, > Thomas

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Voicemail via web interface
by John 24 Apr '18

24 Apr '18

Hello, Can you point me to any additional information about the voice mail via web interface? I have it up and running; however if you click the play button there is no playback, if you click download it will play in MS media player. Thanks John

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mod_conference scalability
by Brian 19 Apr '18

19 Apr '18

Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian.

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WebSocket behind NGINX
by Anton 04 Dec '17

04 Dec '17

Hello All, I have to proxy all websocket requests though a nginx server. Right now I am using next configuration: map $http_upgrade $connection_upgrade { default upgrade; '' close; } server { listen 443; server_name wss.somedomain.com.ua; ssl on; ssl_certificate /etc/nginx/cert.pem; ssl_certificate_key /etc/nginx/private.key; location / { proxy_pass http://127.0.0.1:5066; proxy_http_version 1.1; proxy_set_header Upgrade $http_upgrade; proxy_set_header Connection $connection_upgrade; proxy_read_timeout 86400s; } access_log /var/log/nginx/wss_access; error_log /var/log/nginx/wss_error debug; } I dumped traffic from nginx and found out that "switching protocol" phrase was successful but INVITE message from my browser in pending state. Maybe FreeSWITCH wants real IP not loopback? Who have faced with similar problem? BR, Anton

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Embedding Freeswitch
by Naveen Khanna 23 Nov '17

23 Nov '17

Hi, Can someone suggest a low cost standard single board computer, around $ 50, to run 25 concurrent sessions & 100 registrations of SIP clients with limited applications of Freeswitch. Regards, Naveen Khanna

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FreeSWITCH on Raspberry board
by youssef elouam 20 Sep '17

20 Sep '17

hi all ; I am trying install freeSWITCH in raspberry board (Raspbian OS). but when I run "make && make install" I get as result : make: *** No targets. Stop. could you please help fix this issue. thanks in advance Bestregards --------------------------Youssef ELOUAM.

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FreeSWITCH Week in Review (Master Branch) December 5th – December 12th
by Ken Rice 15 Jun '17

15 Jun '17

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1P8dyo3 FreeSWITCH Week in Review (Master Branch) December 5th – December 12th This week the verto communicator had some new updates to the administrator menu and the core added a new origination_audio_mode variable. Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! This week we have Italo Rossi and the Evolux call center team! And, head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-8616 [verto_communicator] A new menu for moderator, added gain buttons, and removed the 3-dot-button, moving its behavior to member-name div FS-8632 [core] Add origination_audio_mode originate variable with options for sendonly, recvonly or sendrecv Improvements in build system, cross platform support, and packaging: FS-8293 [verto] Add quality level 0 to conference (default is 1) and fix some logic in auto bandwidth The following bugs were squashed: FS-8625 [core] Fixed a segfault caused by an external incoming call from Google Voice. FS-8642 [core] Fixed CF_VIDEO_READY being set on non-video calls FS-8603 [verto_communicator] Added device validation to prevent lost microphones after reset FS-8640 [verto_communicator] Don’t clear conference member reservation id on members that don’t have a reservation ID FS-8633 [mod_verto] Fix for the first verto to join a conference does not get “conference-livearray-join” event FS-8621 [mod_av] Fixed H264 HD1080P video quality issues FS-8631 [mod_db] Updated the regex to allow DSN to match the rest of FS code FS-8643 [mod_sofia] Fixed some memory leaks

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Re: [Freeswitch-users] Fwd: hook to event or startup script which will hook to events?
by Mimiko 27 Mar '17

27 Mar '17

On 14.02.2016 02:38, Abdul Hakeem wrote: > Hello Mimiko, > Did you find any resolution to the issue ? > Cheers, > AH No, no answers to my questions. I use start-up script in lua.conf and bind to event in order to avoid segfault. -- Mimiko desu.

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Memory leak
by Sergey Safarov 30 Dec '16

30 Dec '16

I use monitoring tool (zabbix) to check health of FreeSwitch servers. I has found that after 60 days of uptime FS host has less free memory than after server reboot. This memory is returned back after FS daemon restart. On FS host is installed 66fe6e5. How to correctly detect memory leak in FS?

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