April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Cron-like execution in FS
by Matthew Fong 10 Feb '21

10 Feb '21

I'm wondering if there's any features that allow the cron-like execution of code inside of Freeswitch, preferably with lua--or if I am stuck using the api interface and running the cron outside of freeswitch. --matt

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answer command
by Maxim Tsvetov 21 Sep '20

21 Sep '20

Hello, I found in WIKI list of supported commands http://wiki.freeswitch.org/wiki/Mod_commands. I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32) That's what Freeswitch replies for this command: "answer: Command not found!" Сould you please help? Regards, Maxim Tsvetov

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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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create dummy channel to record conference
by Alexandre Mouillé 07 Oct '14

07 Oct '14

Hi, I hope I'm not posting this in the wrong mailing list. I'm working on a custom conference that uses the conference module. I need a record function that keeps recording (or at list appends the same wav file) when no one is in a given conference. The solution that was used in the Asterisk program I'm porting was to create a dummy channel that goes in the conference and records what it hears in a file. - Can I create a dummy channel with an "originate /loopback" that would go into a conf and record what it hears (I'm not sure what loopback really is) - Otherwise, can I use originate from the FS box to call itself, one side connects to the conference and the other records. (I'm not sure how to do that, and how I could transmit variables (to tell which file to record to) to the called program) Thanks

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Dingaling In Component Mode "How To Call JingleTo SIP"
by Ali Jawad 11 Aug '14

11 Aug '14

Dear All This might be a duplicate post, I am sorry if it is. Normally I get direct feedback from the list. But this time I did not get any reply. So I was afraid it might not have been delivered. First I would like to state that dingaling is working in client mode with gtalk perfectly. I can call from gtalk to my sip extension "1000". I have setup component mode now I am using FS with Openfire, open does show the following: 1 fs.jabber2.mydomain.net LibDingaLing gateway However stats on the jabber server are Session Statistics: Packets Received/Sent: 1/1 So the component is REGISTERED, so far so good my problem now is how to initiate a call from XMPP to a SIP extension. Using client mode I would do the following: 1. Setup GTALK account and assign extension to it 2. Login with another gtalk account and dial the first gtalk account 3. The assigned extension rings I don't know how to do the same using component mode. Adding sip+1000@mydomain to the jabber client "Jabbin" does not popup an "add contact" on XTEN. Anything I missed here ? Thanks for any hints or suggestions The Freeswitch server logs are below: 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" to="fs" version="1.0"></stream:stream> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:component:accept" from="fs.jabber2.MyDomain" id="69fcaa7b"></stream:stream> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <handshake>72dac833190326bb82dddedae558b2b794df5e1b</handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <handshake></handshake> 2008-04-11 13:57:32 [INFO] libdingaling.c:1274 on_log() RECV: ------------------------------------------------------------------------ ------- <iq type="get" id="970-504" from="component.jabber2.MyDomain" to="fs.jabber2.MyDomain"> <query xmlns="http://jabber.org/protocol/disco#info"></query> </iq> 2008-04-11 13:57:32 [NOTICE] libdingaling.c:1276 on_log() SEND: ------------------------------------------------------------------------ ------- <iq from="fs" to="component.jabber2.MyDomain" id="970-504" type="result"> <query xmlns="http://jabber.org/protocol/disco#info"> <identity category="gateway" name="LibDingaLing"></identity> <feature var="http://jabber.org/protocol/disco"></feature> <feature var="http://jabber.org/protocol/disco#info"></feature> <feature var="jabber:iq:version"></feature> <feature var="vcard-temp"></feature> <feature var="http://www.google.com/xmpp/protocol/voice/v1"></feature> <feature var="jabber:iq:last"></feature> </query> </iq>

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TLS Audiocodes
by Szasz Szabolcs 26 Dec '13

26 Dec '13

Hi, Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got to work TLS between freeswitch and a softphone (phonerlite), but I have problem with Audiocodes during the TLS authentication. I've loaded the certification but it still doesn't work. Can I debug the tls in freeswitch? Please help me. Thank you in advance. Szasz Szabolcs

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Is it necessary to call hangup....
by Moiz Chinoy 07 Nov '13

07 Nov '13

Hello, Is it necessary to call hangup in dialplan after say playing a file through playback application. -- Regards, Moiz Chinoy.

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Newbie's installation: Can't start FS at boot
by Dyna Guy 23 May '13

23 May '13

I am a newbie to FS and I want learn it. So I setup a Centos 5.2 (fresh install) and followed the instruction on the wiki installed the new Freeswitch v1.0.3. After the installation I can start FS by issus command "/usr/local/freeswitch/bin/freeswitch". After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch and chmod it to 755. Reboot the server and FS is not running. If I try the start up script: [root@localhost build]# /etc/init.d/freeswitch start Starting freeswitch: [ OK ] [root@localhost build]# /etc/init.d/freeswitch status freeswitch dead but subsys locked Did I miss somthing here? Please help. Thanks a lot. dynaguy Here is the copy of the origenal freeswitch.init.centos5 ---------------------------- [root@localhost ~]# cat /etc/init.d/freeswitch #!/bin/bash # # /etc/rc.d/init.d/freeswitch # # The FreeSwitch Open Source Voice Platform # # chkconfig: 345 89 14 # description: Starts and stops the freeswitch server daemon # processname: freeswitch # config: /usr/local/freeswitch/conf/freeswitch.conf # pidfile: /usr/local/freeswitch/log/freeswitch.pid # # Source function library. . /etc/init.d/functions PROG_NAME=freeswitch PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} FS_USER=${FS_USER-freeswitch} FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} FS_HOME=${FS_HOME-/usr/local/freeswitch} LOCK_FILE=/var/lock/subsys/freeswitch FREESWITCH_ARGS="-nc" RETVAL=0 # Source options file if [ -f /etc/sysconfig/freeswitch ]; then . /etc/sysconfig/freeswitch fi # <define any local shell functions used by the code that follows> start() { echo -n "Starting $PROG_NAME: " if [ -e $LOCK_FILE ]; then if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then echo echo -n $"$PROG_NAME is already running."; failure $"$PROG_NAME is already running."; echo return 1 fi fi cd $FS_HOME daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE $FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1" echo RETVAL=$? [ $RETVAL -eq 0 ] && touch $LOCK_FILE; echo return $RETVAL } stop() { echo -n "Shutting down $PROG_NAME: " if [ ! -e $LOCK_FILE ]; then echo echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not running." failure $"cannot stop $PROG_NAME: $PROG_NAME is not running." echo return 1; fi cd $FS_HOME $FS_FILE -stop > /dev/null 2>&1 killproc $PROG_NAME RETVAL=$? echo [ $RETVAL -eq 0 ] && rm -f $LOCK_FILE; return $RETVAL } rhstatus() { status $PROG_NAME; } case "$1" in start) start ;; stop) stop ;; status) status $PROG_NAME RETVAL=$? ;; restart) stop start ;; reload) # <cause the service configuration to be reread, either with # kill -HUP or by restarting the daemons, in a manner similar # to restart above> ;; condrestart) [ -f $PID_FILE ] && restart || : ;; *) echo "Usage: $PROG_NAME {start|stop|status|reload|restart}" exit 1 ;; esac exit $RETVAL

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