July 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Stumped by mod_portaudio permissions issue
by Mark Haun 02 Feb '20

02 Feb '20

I built freeswitch version 1.7.0+git~20150730T192909Z~ab7f83c654~32bit for my ARM-based home server (an NVIDIA Jetson board), with the intention of using mod_portaudio with a wireless USB headset for my main home phone. It works well when running freeswitch directly as root, but when I configured freeswitch to run at system startup, mod_portaudio could no longer find any audio devices. No problem, I thought, it's a simple permissions issue. I am using the example debian init script from the wiki (this is a Jessie install with SysV init rather than systemd). It runs freeswitch as user freeswitch, group daemon using the -u and -g command-line options. So I added "freeswitch" to the audio group, then used sudo to verify that I could play sound files with "aplay" as the freeswitch user. That worked, but it did not solve the mod_portaudio failure. Thinking there must be an issue with portaudio separate from ALSA, I fetched and built the "pa_devs" utility which enumerates and prints the available audio devices. (This is basically what mod_portaudio is doing on startup.) Running as the freeswitch user, that worked too. Hmmmm. I tried running freeswitch from the command line using sudo rather than the -u and -g options, i.e. $ sudo freeswitch /usr/local/freeswitch/bin/freeswitch -nc and that worked as well! Then I started experimenting with -u and -g and discovered that $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch allows mod_portaudio to load correctly, whereas $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch -g daemon causes mod_portaudio to fail. Can anyone explain what's happening here? In both cases the process UID, GID, EUID, EGID, FUID, and FGID are the same (freeswitch:daemon) as displayed with "ps xao pid,uid,gid,euid,egid,fuid,fgid". Could this be a bug in freeswitch? There are a couple of threads from 2-3 years ago in the archives, where mod_portaudio permissions issues were never resolved and the solution remained a mystery. For the time being I guess this (not using -g) is a workaround, but I'd sure like to understand what is going on. Mark

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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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Set b-leg CallerID using from in a 300 Multiple Choices reply
by Joseph Waite 12 Feb '18

12 Feb '18

Hi Guys I am trying to use the From field in a 300 Multiple Choices message to set the outbound CallerID in the re-directed b-leg So call comes into FreeSwitch, gets sent out to a SIP Redirect server which replies with a 300 Multiple choices. I need to pull the number from the From field of this 300 Response and use it to set the From field, raid & Paid fields in the re-directed invite. Any help would be very much appreciated. Regards

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Received an unsupported RTCP packet version 3
by Mikhail Demekhov 16 Jan '18

16 Jan '18

Hello! I have FreeSWITCH Version 1.6.12. After connecting partner FS-1.7 I had a lot of warning: [WARNING] switch_rtp.c:6207 Received an unsupported RTCP packet version 3 How to fix it? -- Regards, Mikhail Demekhov

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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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Options for initiating call from browser (not WebRTC)
by Oivvio Polite 14 Nov '17

14 Nov '17

I have a simple call recording solution set up with a Twilio-like PaaS. The user interacts with a web app where she enters a destination number. My web app then instructs the PaaS to 1. Call the users phone 2. Wait for the user to answer 3. Start a recording. 4. Connect the destination number. The resulting wav is available for the user to download in the web app. I now want to replace the PaaS with FreeSwitch. I've done a lot of experimenting with FS during the last year, and managed to get callrecording of PSTN and WebRTC calls to work. With those calls origination was from PSTN or WebRTC. Now I want to do origination with a HTTP request. When leafing through the FreeSwitch book from PACKT I see that I have multiple options for doing this. 1. fs_cli Have the browser talk to a separate web app (written in node or python or whatever) that the initiates the call with fs_cli. 2. mod_httapi Have the browser talk to a webapp that talks to FS via mod_httapi. In this scenario I can't find any info on how to originate calls in mod_httapi though. 3. mod_event_socket Have a the browser talk to a webapp that talks to FS via mod_event_socket. In all of these scenarios I figure that I'll have the webapp running on the same box as FS and that I'll do all authentication/authorization of the user in the webapp. But which option should I go with? regards Oivvio

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