October 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_lua problems: /usr/local/lib/lua/5.1/genx.so: Undefined symbol "lua_call"
by Борисов, Дмитрий 08 Oct '14

08 Oct '14

I have a myself writed script, it must read data from database and create XML config for FS. For creating XML I using genx library, it was installed by luarocks. This library worked fine when I use it in standalone scripts, but when I use it in FS I`ve got next error (I`ve put there bigger part of freeswitch.log for better explanation): 2014-10-07 15:34:02.010800 [INFO] switch_cpp.cpp:1328 package.path /usr/local/share/freeswitch/scripts/lua/?.lua package.cpath /usr/local/lib/lua/5.1/?.so;/usr/local/lib/?.so 2014-10-07 15:34:02.010848 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, provided params: 'Event-Name: GENERAL Core-UUID: ed5920d2-154e-e411-b150-005056a83c1f FreeSWITCH-Hostname: voip-bsd-01.local FreeSWITCH-Switchname: voip-bsd-01.local FreeSWITCH-IPv4: xxx.xxx.xxx.xxx FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2014-10-07%2015%3A34%3A02 Event-Date-GMT: Tue,%2007%20Oct%202014%2011%3A34%3A02%20GMT Event-Date-Timestamp: 1412681642009651 Event-Calling-File: switch_core.c Event-Calling-Function: switch_load_network_lists Event-Calling-Line-Number: 1393 Event-Sequence: 397 domain: company1.xxxx.com purpose: network-list ' 2014-10-07 15:34:02.010858 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua: XML_REQUEST[section]:directory XML_REQUEST["tag_name"]:domain XML_REQUEST["key_name"]:name XML_REQUEST["key_value"]:company1.xxxx.com 2014-10-07 15:34:02.016535 [DEBUG] freeswitch_lua.cpp:360 DBH handle 0x80c32d0f0 Connected. 2014-10-07 15:34:02.016916 [ERR] mod_lua.cpp:203 error loading module 'genx' from file '/usr/local/lib/lua/5.1/genx.so': /usr/local/lib/lua/5.1/genx.so: Undefined symbol "lua_call" stack traceback: [C]: in ? [C]: in function 'require' .../local/share/freeswitch/scripts/lua/gen_dir_user_xml.lua:202: in main chunk 2014-10-07 15:34:02.016923 [ERR] mod_lua.cpp:269 LUA script parse/execute error! My lua.conf.xml: <configuration name="lua.conf" description="LUA Configuration"> <settings> <param name="module-directory" value="/usr/local/lib/lua/5.1/?.so"/> <param name="module-directory" value="/usr/local/lib/?.so"/> <param name="script-directory" value="$${script_dir}/lua/?.lua"/> <param name="xml-handler-script" value="lua/gen_dir_user_xml.lua"/> <param name="xml-handler-bindings" value="directory"/> </settings> </configuration> I`d attached script because it is large. Any thinks about methods for solving problem? -- best regards, Dmitriy Borisov

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Multiple features sought
by Will Stewart 08 Oct '14

08 Oct '14

I'm interested in Freeswitch because I'd like to incorporate the following into my home automation system; - Whole house audio: home runs from a central patch panel are acceptable - Play wake up sounds (surf, jungle, etc) in individual bedrooms - Whole house intercom Is anyone doing some or all of the above with Freeswitch? If not, is it possible/practical? Thanks, Will

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LDAP
by Nick Zaitsev 08 Oct '14

08 Oct '14

Good day to you. Could you advise me how i can implement auth users via LDAP in the simplest way? Do i have to write my own application for mod_xml_curl(maybe there is such application somewhere)? Or i can use the mod for freeswitch and it will work like a charm.(i use both sip and skinny users) Thank you for your time, Best Regards Nick Zaitsev

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local extensions <> DID
by Jerry Allen 08 Oct '14

08 Oct '14

For a public voip server, I would like to use the 10 digit DID for the extension and be able to route all non-local DID out to my gateway(s). Is it possible to use the PSTN DID as the SIP extension? I do not find any information concerning this. I understand DID > extension as long as extension matches the predefined <extension name="Local_Extension"> destination_number expression. The issue is using a 10 digit extension collides with the 10 digit routing to my gateway and loops. Jerry Allen

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Audio issues
by Andrew 08 Oct '14

08 Oct '14

Hi, I have a very odd issues and looking for some help. I am using Master FS and had this issue before. I am making an outbound call from my program to my cell and playing an audio file. I am talking back in my phone to the audio file and found that The more I talked the more the audio breaks up. I stop talking and then the audio stops playing. I start talking and the audio starts playing where it left off. I am sure its not a packet loss as the audio file continues where it stop playing. I also have FS client open and do not see any errors. I can 100% reproduce this issue. I did try 3 ways to play the audio and different audio files. session.read session.PlayAndGetDigits session.StreamFile I tried a direct connect to me to a cisco switch and other voip routes. Anyone know if this is a bug maybe locking thread or a setting? Andrew

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Transferring a PSTN call via gateway to an internal user
by Chirag Desai 07 Oct '14

07 Oct '14

Hi, I have users registering to Kamailio and calls flowing to Freeswitch. I have a multi-domain set up. Users are only able to call other users within their domain. I'm able to place calls with no issue between internal users on the same domain e.g. 1000 to 1019. Users are also able to call externally via an external gateway (sip provider). My problem arises when transferring calls, either blind or attended. What's the best practice here? Should the call when transferred hit the public.xml dialplan or the default.xml dialplan? I'm guessing this is defined in the "context" for the gateway. And once they're in the dialplan do I route them back out to kamailio then back to FS? Or should FS find the user and try to connect the call. I managed to get as far as making an oubound call, transferring (blind) to another internal user. But upon transfer there is no sound either way. (I had to hard code the domain to make this work). I've been stuck here for a couple weeks now and would really appreciate some help. Thanks in advance for your help, C

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Transferring a PSTN call (via gateway) to an internal user
by Chirag Desai 07 Oct '14

07 Oct '14

Hi all, I have users registering to Kamailio and calls flowing to Freeswitch. I have a multi-domain set up. Users are only able to call other users within their domain. I'm able to place calls with no issue between internal users on the same domain e.g. 1000 to 1019. Users are also able to call externally via an external gateway (sip provider). My problem arises when transferring calls, either blind or attended. What's the best practice here? Should the call when transferred hit the public.xml dialplan or the default.xml dialplan? I'm guessing this is defined in the "context" for the gateway. And once they're in the dialplan do I route them back out to kamailio then back to FS? Or should FS find the user and try to connect the call. I managed to get as far as making an oubound call, transferring (blind) to another internal user. But upon transfer there is no sound either way. (I had to hard code the domain to make this work). I've been stuck here for a couple weeks now and would really appreciate some help. Thanks in advance for your help, C

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Playing on hold music for certain inbound numbers
by Tom Angle 07 Oct '14

07 Oct '14

I am new to Freeswitch. I am trying to create a MOH server for our Metaswitch. Currently I have it working, but it will only play on audio file. I would like to set it up to play a particular audio depending on the inbound phone number. The below code is in my dialplan/default.xml and does not work. Could someone tell me what I am doing wrong? Thanks for any help. <extension name="4000_hold_music"> <condition field="effective_caller_id_number" expression="^(1000)$"> <action application="answer"/> <action application="execute_extension" data="is_secure XML features"/> <action application="playback" data="$${hold_music}"/> <anti-action application="set" data="zrtp_secure_media=true"/> <anti-action application="answer"/> <anti-action application="execute_extension" data="is_zrtp_secure XML features"/> <anti-action application="playback" data="/music/music1.wav"/> </condition> <condition field="effective_caller_id_number" expression="^(1005)$"> <action application="answer"/> <action application="execute_extension" data="is_secure XML features"/> <action application="playback" data="$${hold_music}"/> <anti-action application="set" data="zrtp_secure_media=true"/> <anti-action application="answer"/> <anti-action application="execute_extension" data="is_zrtp_secure XML features"/> <anti-action application="playback" data="/music/music4.wav"/> </condition> </extension>

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fs_path with a 'transport' parameter
by Yuriy Nasida 07 Oct '14

07 Oct '14

Hello guys, Looks like it is not possible for now but may be nice to have. This way we will able to send INVITE from FS to opensips via UDP and let opensips know that it should use TLS for INVITE to endpoint. Any thoughts ? Thanks.

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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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