May 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Any reason why this Freeswitch init script fails to start (after modifications)
by vfclists . 02 May '16

02 May '16

This is the pastebin of the freeswitch init script on my Debian 8 system. http://pastebin.com/7mmw9ghx Freeswitch starts OK, only that it starts with the wrong configuration, ie I can log in via fs_cli but then non of the conf files would have kicked. Even when I disable it and run freeswitch from /etc/rc.local Freeswitch doesn't work. I usually have to login first then run Freeswitch from the command line, but is not very good because if the system needs to be rebooted for some reason Freeswitch needs to be manually started. When I examine the file the only thing that seems to be wrong is that the DAEMON_ARGS variable does not include the freeswitch configuration file or directory or that the configuratin file defined by /etc/default/$NAME is undefined. This seems to imply that if I add " -c /usr/local/freeswitch/conf' " to DAEMON_ARGS freeswitch should start but it doesn't. If that is the case what should go into /etc/default/$NAME ? # PATH should only include /usr/* if it runs after the mountnfs.sh script PATH=/sbin:/usr/sbin:/bin:/usr/bin:/usr/local/bin DESC="Freeswitch" NAME=freeswitch DAEMON=/usr/local/freeswitch/bin/$NAME DAEMON_ARGS="-nc -nonat" PIDFILE=/usr/local/freeswitch/run/$NAME.pid SCRIPTNAME=/etc/init.d/$NAME FS_USER=freeswitch FS_GROUP=daemon # Exit if the package is not installed [ -x "$DAEMON" ] || exit 0 # Read configuration variable file if it is present [ -r /etc/default/$NAME ] && . /etc/default/$NAME # Load the VERBOSE setting and other rcS variables . /lib/init/vars.sh # Define LSB log_* functions. # Depend on lsb-base (>= 3.0-6) to ensure that this file is present. . /lib/lsb/init-functions -- Frank Church ======================= http://devblog.brahmancreations.com

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Multiple inbound numbers on a registered trunk
by Joseph Waite 02 May '16

02 May '16

Hi Guys We have a Freeswitch based FusionPBX setup which registers to our main SIP Switch with Registering trunks. Outbound Calls work fine, however I’m having issues with using multiple inbound numbers. If I setup a destination with the username of the trunk then all calls received on the trunk goto that destination. If I setup each DID on its own then calls fails. In the invite I see To: <sip:trunkuser@providerip:5080> I see the DID dialled in the following in the SIP message Diversion: numberdialed <tel:numberdialed>;reason=unknown;counter=1;privacy=off Is there a setting I can add somewhere to the trunk config to tell it to use the value in Diversion? Any help greatly appreciated. Regards Joseph Waite

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SIP cause codes in CDR
by Dmitry Mordovin 02 May '16

02 May '16

Hello My VoIP provider sometimes refuse my call, for example, with 487 message (ORIGINATOR_CANCEL) or 503 (NORMAL_TEMPORARY_FAILURE) Does anyone know the way (channel variable, function call) how can I put to CDR record 487, 503 and other SIP notation cause number instead Q.850? The main reason, SIP equivalents representation simplify troubleshooting work. DM

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What are current best options for WebRTC between mobile and FreeSwitch
by Oivvio Polite 01 May '16

01 May '16

I'm working on a project that will need desktop as well as mobile clients. On the desktop side I'm using Electron[1]. On mobile I'm targeting Android and iOS but I haven't picked the development tools yet. The mobile client is a lot simpler than the desktop and on recent versions of Android I hope to get away with using Chrome. But for older versions of Android and for iOS I'll need an app. My main candidates there PhoneGap/Cordova[2] and React Native[3]. React Native seems to have a lot of promise but PhoneGap has more of a track record and there seems to be people actually using it in production for WebRTC. Since stability is really important to this project I'm leaning towards PhoneGap/Cordova. All of these tools use a Javascript runtime so I hope to be able to use verto.js for my signalling regardless of what I choose. But I'm curious about the experiences you folks have with these tools. Oivvio [1] http://electron.atom.io/ [2] https://webrtchacks.com/webrtc-hybrid-applications/ [3] https://webrtchacks.com/reacting-to-react-native-for-native-webrtc-apps-ale…

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Problem with Playback wav from remote server
by Pete Kay 01 May '16

01 May '16

Hi I am able to playback a remote mp3 but not a remote .wav file. I am getting strange error. Could someone please guide me how to resolve it? freeswitch@server3> 2016-05-01 10:37:07.751782 [DEBUG] mod_httapi.c:2631 caching: url:http://105.165.2.110/voice/m79eVzDuH6_61.wav to /usr/local/freeswitch/storage/http_file_cache/c8691a860ba4ddc0b18a39d3768add75.wav (5773 bytes) 2016-05-01 10:37:07.771763 [ERR] mod_sndfile.c:204 Error Opening File [/usr/local/freeswitch/storage/http_file_cache/c8691a860ba4ddc0b18a39d3768add75.wav] [File contains data in an unknown format.] 2016-05-01 10:37:07.771763 [ERR] mod_httapi.c:2971 Invalid cache file /usr/local/freeswitch/storage/http_file_cache/c8691a860ba4ddc0b18a39d3768add75.wav opening url 105.165.2.110/voice/m79eVzDuH6_61.wav Discarding file. 2016-05-01 10:37:07.771763 [DEBUG] sofia.c:6755 Channel sofia/internal/ 0000000000(a)105.165.2.110 entering state [early][183] Here it shows that I can "get" the .wav no problem. [root@ server3 ~]# wget http://105.165.2.110/voice/m79eVzDuH6_61.wav --2016-05-01 10:15:42-- http://105.165.2.110/voice/m79eVzDuH6_61.wav Connecting to 105.165.2.110:80... connected. HTTP request sent, awaiting response... 302 Found Location: http://105.165.2.110/auth/login [following] --2016-05-01 10:15:42-- http://105.165.2.110/auth/login Connecting to 105.165.2.110:80... connected. HTTP request sent, awaiting response... 200 OK Length: 5773 (5.6K) [text/html] Saving to: `m79eVzDuH6_61.wav' 100%[================================================================================================================================================>] 5,773 --.-K/s in 0s 2016-05-01 10:15:42 (483 MB/s) - `m79eVzDuH6_61.wav' saved [5773/5773] [root@ server3 ~]# ls -al *.wav -rw-r--r-- 1 root root 5773 May 1 10:15 m79eVzDuH6_61.wav [root@server3 ~]#

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Recording video and timing
by Luca Pradovera 01 May '16

01 May '16

Hello, after a long journey I finally managed to get my "one presenter + multiple Verto clients" setup. However, I think I might have hit some bug in the recording code. Recordings usually work well for the first 2-3 minutes, then I get a few freeze frames. After that, video is out of sync with audio and is recorded and played back at roughly 3x the correct speed. I am using the "transcode" video mode with VP8, and there seems to not be any error in the console. Where can I start debugging such an issue? Best regards, Luca Pradovera

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