April 2009 - 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

5 5

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

7 15

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>

4 8

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

3 2

How to break a playback with loops
by Daniel Liang 24 Jun '12

24 Jun '12

Hi, I am trying to playing back a wav file a few times, and on receiving a dtmf event, it stops playing the file. The command I sent is: sendmsg call-command: execute execute-app-name: playback execute-app-arg: <the wav file> loops: 4 On receiving a dtmf event, I send a break command: sendmsg call-command: execute execute-app-name: break However, that only breaks one loop of the playback. And I have to press a dtmf for 4 times to stop the file. I tried sending 4 break commands all at once. But it didn't work. And even worse, my program stops receiving dtmfs anymore. My guess is that the first break stops the first loop, but before the second loop starts, the following 3 breaks stop something else that I don't know what they are. Any help is appreciated. Thanks. Daniel Liang

6 6

Grandstream GXP 2000
by Jonas Gauffin 18 Dec '11

18 Dec '11

Hello I have problems with TCP and GXP2000 phones. They work fine using UDP, but no calls arrive if I switch to TCP. I've upgraded to the latest GS firmware today to see it that helped, but it didn't. Have anyone else had problems with TCP? Regards, Jonas

5 8

xml_curl and gateways
by Peter P GMX 21 Feb '11

21 Feb '11

According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to set up gateways dynamically, but I do net get it to work: I always get : "Invalid profile" My assumptions for a right xml answer back to FS are as follows, but I think at least one of it is false: 1. ) I start with <document type='freeswitch/xml'> <section name='configuration'> <configuration name='sofia.conf' description='RailsAdditionalConf'> 2.) I do not want to change settings, so I do NOT add a <settings> tag 3.) I want to enhance an existing profile, so I add a profile tag (<profile name='external'>) 4.) I assume that I can have an existing profile "external" in the xml conf files and can mix additional parts dynamically into the same profile 5.) Within the profile I want to add gateways, so I add a gateway tag (<gateways>) 6.) Then I add the gateways 7.) I add the closing tags Which one of the assumptions if wrong? The request I receive is as follows: {"key_name"=>"name", "action"=>"directory", "tag_name"=>"configuration", "key_value"=>"sofia.conf", "controller"=>"xml_curls", "hostname"=>"freeswitch", "section"=>"configuration"} I only answer once at startup with the follwing: <document type='freeswitch/xml'> <section name='configuration'> <configuration name='sofia.conf' description='RailsAdditionalConf'> <profile name='external'> <gateways> <gateway name="freeswitch2"> <param name="username" value="1007"/> <param name="realm" value="192.168.0.56:5061"/> <param name="password" value="xxxx"/> <param name="proxy" value="192.168.0.56"/> <param name="expire-seconds" value="30"/> <param name="register" value="true"/> <param name="register-transport" value="tls"/> <param name="retry-seconds" value="30"/> </gateway> </gateways> </profile> </configuration> </section> </document> Does anybody have a correct XML answer for a gateway and can post it? Best regards Peter

6 11

Stream audio file/live to multiple SIP endpoints with IP multicast
by Sluschny, Thomas 30 Aug '10

30 Aug '10

Hi, i want to stream audio (from file and live) from FreeSwitch to multiple SIP endpoints (these are proprietary). Because of bandwith limitations we have to use IP multicast for audio data. Now the question: how can i do this smartly in FreeSwitch? I have read some Wiki entries: - mod_esf: seams to use multicast, but i couldn't find usage information - mod_shout: streaming, but from internet ? MP3, but i want wav-files to stream - mod_local_stream: seams to play files in a loop, i want to play a whole file once. - mod_conference: seams not to support IP multicast Im searching for a simple way to send a SIP INVITE to all my endpoints and then stream audio (live from soundcard or from file) over IP multicast to all these endpoints. Thanks in advance, Thomas

3 3

Direct inward dialling
by Peter P GMX 07 Jun '10

07 Jun '10

Dear all, when I get an incoming call from my SIP provider I do receive an invite on the trunk number, e.g. in Germany 0xxxxx9. However I have an extension block of 0xxxxx90 to 0xxxxx999. In the dialplan I checked the condition field="destination_number" but this compares to the number of the trunk 0xxxxx9, so I do not get the direct inward dialling extensions. The SIP to-header contains the direct inward dialling number 0xxxxx90 to 0xxxxx999 but I have no success in accessing it in the dialplan. I tried sip_to_user, sip_to_uri, sip_req_user, sip_req_uri, but no success. They are all empty. Any idea which variable to parse? Best regards Peter

10 29

OSP Interop w/ Trans Nexus
by Gregory Boehnlein 02 Dec '09

02 Dec '09

Anyone know if Freeswitch can work w/ OSP and use something like Transnexus for CDR/Rating?

6 5