March 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Unable to load mod_java.
by John 21 Apr '15

21 Apr '15

Hi All I have built mod_java and have mod_java.so in /usr/local/freeswitch/mod/ but when I am trying to load mod_java from freeswitch terminal like : load mod_java getting following error : 2015-02-07 15:01:13.175331 [INFO] mod_enum.c:880 ENUM Reloaded 2015-02-07 15:01:13.175331 [NOTICE] modjava.c:366 Java Framework Loading... 2015-02-07 15:01:13.175331 [INFO] switch_time.c:1411 Timezone reloaded 1781 definitions +OK Reloading XML -ERR [module load file routine returned an error] 2015-02-07 15:01:13.215310 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** I am using OpenJDK version : java version "1.6.0_33" OpenJDK Runtime Environment (IcedTea6 1.13.5) (6b33-1.13.5-1ubuntu0.14.04) OpenJDK 64-Bit Server VM (build 23.25-b01, mixed mode) and my conf/autoload_configs/java. conf.xml reads as : <configuration name="java.conf" description="Java Plug-Ins"> <javavm path="/usr/lib/jvm/java-6- openjdk-amd64/jre/lib/amd64/jamvm/libjvm.so"/> <options> <option value="-Djava.class.path=/usr/local/freeswitch/scripts/freeswitch.jar"/> <option value="-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=0.0.0.0:8000"/> </options> <startup class="org/freeswitch/example/ApplicationLauncher" method="startup"/> </configuration> Can someone help me to resolve this error? I tried googling this issue and accordingly made above config changes. Thanks, John

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Set channel variables before bridge leg B hangup
by Zoltán Szabó 14 Apr '15

14 Apr '15

Hi, In lua I bridge two sessions. When leg B hangup the call I need to set up some custom channel variables for odbc_cdr reporting. freeswitch.bridge(session1, session2); session2:execute("set", "custom_var1=asdf"); But when the set command tries to run, the log says "channel is hangup already". Is there any way to do this properly? Many thanks, Zoltan

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Intermittent audio issue with Chrome 40-42 and WebRTC
by Oleg Stolyar 09 Apr '15

09 Apr '15

Hi guys, Ever since my users switched to Chrome 40, 41 or 42 beta, every now and then they complain that they cannot hear audio from FreeSWITCH or send audio to FreeSWITCH. I cannot reproduce this at will but was able to catch several occurrences. It happens on both Windows and Macs. Here are some details" 1. Usually the problem is solved if we close Chrome and make sure all the Chrome processes are killed. 2. When the problem happens, FS logs this on connection: [ERR] switch_rtp.c:2976 audio DTLS packet not written 3. I tried using this WebRTC test on an affected computer: https://janus.conf.meetecho.com/echotest.html and that worked, so it is not a general WebRTC issue. 4. I tried FS WebRTC demo page https://webrtc.freeswitch.org/verto/index.html#page-main and on the affected computer the problem was still there - no audio. 5. This happened both on an old FS version from last May and on a very recent master FreeSWITCH Version 1.5.15b+git~20150316T164411Z~b32abaadd9~64bit (git b32abaa 2015-03-16 16:44:11Z 64bit) Any ideas?

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Re- Freetdm version inside FreeSWITCH
by Andrew Keil 08 Apr '15

08 Apr '15

To FreeSWITCH users, I am not sure if this should be posted inside the developers mailing list, however I thought I would start here first. Currently I am setting up a FreeSWITCH system (using the current production release version) in the UK which will be running off BT (British Telecom) and use both SIP and ISDN30e. The ISDN side will use Sangoma A104DE cards (which I have just purchased and I am testing currently). Now the fun begins... I came across an Open Circuit alarm issue, which Sangoma are investigating, which then caused an unmapped debug message to happen (correctly from inside FreeSWITCH ftmod_wanpipe.c). Sangoma then suggested I run: ftp://ftp.sangoma.com/linux/custom/freetdm/freetdm-1.9.0.tar.gz {since this has a temporary fix to skip open circuit alarms and let FreeSWITCH continue to run with no DEBUG messages, also it adds a header (ftmod_wanpipe_defines.h) to enable mapping of alarm status values to plain English - instead of the current unmapped message} Which brings me to my primary question: Sangoma have advised me that they manage freetdm and I should use their version. However when I look at the source to ftmod_wanpipe.c (for example) their version (freetdm-1.9.0) has the following header: Copyright (c) 2007-2012, Anthony Minessale II vs. FreeSWITCH current version: Copyright (c) 2007-2014, Anthony Minessale II. Obviously there are other additions/changes done by Sangoma (including the Open Circuit alarm skipping). So what is FreeSWITCH's point of view on this? Do you merge the changes every <x> months from Sangoma's freetdm or simply expect the freetdm component to be overwritten by Sangoma's current version prior to FreeSWITCH being built? My main aim is to have a reasonably well supported system and not come across issues that have already been found and fixed by either Sangoma or FreeSWITCH. Thanks in advance for any feedback on this, as I just want to make sure I am using an agreed best version of freetdm. Andrew Keil Visytel Pty Ltd

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Changing the voicemail recording menu
by Paul Atreides 08 Apr '15

08 Apr '15

Hi how do I change the menu when the voicemail answers for record a new message? I want it to play the - greeting message - record the message - and then hangup after silence Is is possible to change the menu when the user access its voicemail as well? I found the voicemail_ivr.conf.xml, but I cant find any documentation to it in the wiki Thanks

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Early Dial / Sip 484
by Paul Atreides 07 Apr '15

07 Apr '15

Hi, does Freeswitch support Early Dial ( Sip 484 ? ), if yes how do i set it up in the dialplan? Thanks

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Sofia.h limits ?
by Abdul Hakeem 06 Apr '15

06 Apr '15

Hello Guys, Could someone shed lights if these lines are hard codec limits and suggestions for alternative parameters ? #define SOFIA_MAX_AL 100 #define MAX_RTP 50 #define SOFIA_MAX_MSG_QUEUE 64 Many thanks, Abdul Hakeem

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Transfer/shift calls from one freeswitch to other freeswitch box
by Varghese Paul 06 Apr '15

06 Apr '15

Hi all, Is it possible to transfer/forward calls from one freeswitch box to other box ?. Use case: We wanted to do some maintenance on one freeswitch server and thinking to transfer/shift the current calls in one server to other server. Is there any way to route the calls to other freeswitch server without hanging up the call?. I know we should send Re-INVITE to other users for modifying the RTP IP + RTP port number for the new Freeswitch IP. Is it possible to achieve in freeswitch ? Regards Varghese Paul

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Error Mod_lua.
by Antony Aguirre Morales 02 Apr '15

02 Apr '15

Hi, Install fusionpbx to manage freeswitch, perform the configuration of lua module enable these lines in lua.conf.xml <param name = "xml-handler-script" value = "app.lua xml_handler" /> <param name = "xml-handler-bindings" value = "dialplan | directory | configuration" /> I am using the script for the configuration that comes in the pbx in the following path: / var / www / html / fusionpbx / resources / install / scripts copy them to the following path freeswitch: / usr / local / freeswitch / scripts / but when you start freeswitch shows me the following errors have some idea: 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:203 cannot open /usr/local/freeswitch/scripts/resources/config.lua: No such file or directory stack traceback: [C]: in function 'dofile' /usr/local/freeswitch/scripts/app.lua:30: in main chunk 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:269 LUA script parse/execute error! 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:203 cannot open /usr/local/freeswitch/scripts/resources/config.lua: No such file or directory stack traceback: [C]: in function 'dofile' /usr/local/freeswitch/scripts/app.lua:30: in main chunk 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:269 LUA script parse/execute error! 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:203 cannot open /usr/local/freeswitch/scripts/resources/config.lua: No such file or directory stack traceback: [C]: in function 'dofile' /usr/local/freeswitch/scripts/app.lua:30: in main chunk 2015-03-31 07:47:15.386121 [ERR] mod_lua.cpp:269 LUA script parse/execute error! Regards.

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Caller ID prefix missing
by Paul Atreides 01 Apr '15

01 Apr '15

Hi, when I place a call over the gateway to the PTSN I see the dialed number without the leading 0 prefix in the history of my Grandstream GXP2140. Is there a way to add the 0 back again? Thank you <include> <extension name="isdn-outbound"> <condition field="destination_number" expression="^0(\d+)$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/isdn/$1"/> </condition> </extension> </include>

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