September 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Video calls over SIP
by Norman Laidla 08 Sep '14

08 Sep '14

Good day, I've been setting up a video conferencing system in our office. We have a Cisco SX-20 set that we intend to use, but so far, there's been little in the way of progress. In short, video calls don't work. Here's what I'm working with: a Debian server running the latest stable FreeSWITCH build, two laptops with Zoiper and the Cisco SX-20. Audio-only calls work without a hitch, problems arise when I try to start the video stream. The only way to get any kind of image to reach one endpoint from another is to set FS to bypass media and make a video call between my laptop and the SX-20, in which case the outgoing feed from the laptop reaches the SX-20 as a jumbled mess of colors and the occasional still frame. Setting FS to proxy media allows no video to reach the SX-20 and causes a loopback in both laptops. The default setting doesn't do video at all. Funny thing is, if the call is originated from the SX-20, its control panel shows an outgoing video stream, but nothing ever reaches the other endpoint. I'm beginning to think that doing a video conferencing setup with FreeSWITCH is a fool's errand. If anyone has done something of the sort before, I would welcome your advice. Best regards, Norman Laidla

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PCMU codec issue (mismatch?) resulting in Incompatible Destination
by Komar, Jason 07 Sep '14

07 Sep '14

Hi Everyone, I had a Raspberry Pi installation of the latest FreeSWITCH 1.2 branch running on Gentoo. It was working great, but my SD card packed it in. I thought I would try something different, so I loaded up Raspbian and installed FreeSWITCH via Fusionpbx script. I registered my Panasonic KX-TPA50 phone, but when doing a test call to MoH or echo, I get an incompatible destination error. Looks like a codec mismatch. Not sure what to look for though. I presume it's just a setting somewhere. Both FreeSWITCH and the phone have PCMU, but something is different in the comparison as can be seen here: [DEBUG] switch_core_media.c:3475 Audio Codec Compare [PCMU:0:8000:30:32000:1]/[PCMU:0:8000:20:64000:1] I understand the 8000, but I don't know what the other mismatched numbers are or where to look to change a setting to get them to match. The same mismatched numbers show when it compares G722 and PCMA. Hope someone can point me in the right direction. Thanks, Jason

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mod_gsmopen incoming SMS
by Daniel Ivanov 07 Sep '14

07 Sep '14

I receive no event of any sort about incoming SMSes on gsmopen channels. I send and receive calls without a problem but sms messages seems to disappear. Any fast hints on that? mod_sms is loaded and this is my gsmopen.conf : <configuration name="gsmopen.conf" description="GSMopen Configuration"> <global_settings> <param name="debug" value="8"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="hold-music" value="$${moh_uri}"/> <param name="destination" value="9999"/> </global_settings>  <per_interface_settings> <interface id="1" name="XXXX"> <param name="hold-music" value="$${moh_uri}"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="destination" value="5000"/> <param name="imei" value="35XXXXX"/> </interface> <interface id="2" name="XXXXXX"> <param name="hold-music" value="$${moh_uri}"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="destination" value="5000"/> <param name="imei" value="XXXX"/> </interface> <interface id="3" name="XXXXXXX"> <param name="hold-music" value="$${moh_uri}"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="destination" value="5000"/> <param name="imei" value="XXXXXXX"/> </interface> <interface id="4" name="XXXXXXX"> <param name="hold-music" value="$${moh_uri}"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="destination" value="9996"/> <param name="imei" value="XXXXXXXXXX"/> </interface> </per_interface_settings> </configuration>

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Libpri implemented supplementary ISDN services
by babak 07 Sep '14

07 Sep '14

Hi There is some implemented supplementary ISDN services in Libpri need some application to run it from dialplan like Call Deflection.In Asterisk for using this service there is DAHDISendCallreroutingFacility(<destination>, <original-my-number>, cfu|cfb|cfnr|unknown). How we can invoke This service(Call Deflection) in freeswitch ? Regards

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message object isn't defined in chatplan
by Snabel Kabiya 07 Sep '14

07 Sep '14

Hi, I'm following the instructions here to create a script the can intercept chat messages: http://wiki.freeswitch.org/wiki/Mod_sms#send when i send a chat message i get the error bellow, why is that? and why the chat method isn't invoked? 2014-09-07 13:29:53.397788 [INFO] mod_sms.c:336 Processing text message 1111->2222 in context public Chatplan: 2222 parsing [public->public] continue=false Chatplan: 2222(a)x.x.x.1 Regex (PASS) [public] to(2222(a)x.x.x.1) =~ /^(.*)$/ break=on-false Chatplan: 2222(a)x.x.x.1 Action lua(handle_chat.lua) Chatplan: 2222(a)x.x.x.1 Action reply(Hello, you said: ${_body}) Chatplan: 2222(a)x.x.x.1 Action send() *2014-09-07 13:29:53.397788 [ERR] mod_lua.cpp:203 /usr/local/freeswitch/scripts/handle_chat.lua:6: '=' expected near 'body'* *Public.xml:* <?xml version="1.0" encoding="utf-8"?> <include> <context name="public"> <extension name="public"> <condition field="to" expression="^(.*)$"> <action application="lua" data="handle_chat.lua"/> <action application="reply" data="Hello, you said: ${_body}"/> <action application="send"/> </condition> </extension> </context> </include> *handle_chat.lua:* logDebug("handle_chat - start before function"); function chat(message, args) logDebug("handle_chat - start in function"); var body = message:getBody(); logInfo("chatplan body is=" .. body); --message:chat_execute("reply", "you said: " .. message:getBody()); logDebug("handle_chat - end"); end Thanks, Snabel

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there is String function in Dialplan
by matanya cohen 06 Sep '14

06 Sep '14

I need String function in Dialplen, like get substring from String

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play sounds files from dynamic string
by matanya cohen 06 Sep '14

06 Sep '14

I got string from webservice inside Dialplan. This string contain list files with come as separator. I need to play them? Matanya

2 1

Getting to the bottom of a problem
by Alex Lake 05 Sep '14

05 Sep '14

My freeswitch installation (running under Ubuntu 12.04) has a habit of mysteriously stopping. It's not doing this in a predictable manner (eg. it's not when any particular bit of dialplan is being executed). I have a script that runs every minute to see if it's running, and if it isn't, starts it up (and sends me a notification). I'd like to be able to get to the bottom of this, but am not particularly (or at all!) experienced in running gdb. If I follow the brief instructions in the Wiki to run "gdb bin/freeswitch" I don't seem to get a working freeswitch running at all. What should I see on the command line when I issue that command? This what I currently see: ----- root@hzhmay14a /home/oev/fs-pabx/scripts # gdb --args /usr/local/freeswitch/bin/freeswitch -nonat GNU gdb (Ubuntu/Linaro 7.4-2012.04-0ubuntu2.1) 7.4-2012.04 Copyright (C) 2012 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-linux-gnu". For bug reporting instructions, please see: <http://bugs.launchpad.net/gdb-linaro/>... Reading symbols from /usr/local/freeswitch/bin/freeswitch...done. (gdb) (gdb) ----- There's a mysterious (to me) command line parameter "-core" that I was wondering if it might be useful in my circumstances. I'm also prepared to do a new build (with some kind of diagnostic params) if that helps. Any tips or suggestions of things to read up on most welcome....

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inbound-bypass-media
by Ahmed habiba 05 Sep '14

05 Sep '14

Hi, I'm trying to use "inbound-bypass-media=true" in sip profile internal.xml, however when I do this I can not get any voice on caller side, considering that both devices are behind a NAT(in the same private network), and my FreeSWITCH has a public IP as below. [FreeSWITCH]-->internet--->{NAT}--->Phone1 |--->Phone2 Your kind help will be appreciated. -- Thanks and Best Regards, Ahmed Habiba

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FreeSWITCH Friday BugHunt Reminder!
by Ken Rice 05 Sep '14

05 Sep '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken http://ift.tt/vGSsGm http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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