- FreeSWITCH-users - FreeSWITCH Mailing Lists

How to create a new leg on conference in mute using socket interface
by lndspereira-fs＠yahoo.com 08 Mar '13

08 Mar '13

Hi. I would like to know if there is a way to add a new muted member to a conference. I'm using the following command: api conference my_conf_id dial {origination_caller_id_number=sofia/external/+1444123456@12.12.12.12}sofia/external/+1111333444@12.12.12.12 I could not find a way to set the '+flag{"mute"}' in the command above. Thanks in advance, Leo Leonardo Nogueira de Sá Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77

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Quality ANALOG Speaker Phone
by Sean Devoy 08 Mar '13

08 Mar '13

Hi All, Can anyone suggest a true quality analog line speaker phone? My mother has severe hearing loss and cannot use a handset to talk at all. She does "ok" with speaker phone, but her's is terrible quality. All my searches keep up with digital (VOIP) or USB speakerphones. Why not digital, you say? Because they are OLD and stubborn, and of course their house is wired for analog phones! Sean

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Delete X-FS-Support header
by Mickael Hubert 08 Mar '13

08 Mar '13

Hi list, I want delete X-FS-Support header in INVITE message. But this command "<action application="bridge" data="* {ignore_display_updates=true}*sofia/gateway/provider/18005551212"/>" is not OK Have you a other command (or any things else) to delete this header please ? Thanks -- Cordialement Hubert Mickaël Hexanet --

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Re: [Freeswitch-users] BUG??? Bad mixed Audio record_session
by Jesus Ramon Rocha Velazquez 08 Mar '13

08 Mar '13

It happens to 40% of my long duration calls, when it has more than 10 minutes there always the Leg A get out of phase. Is not that the begining gets copies, it's that the last minutes of the call are one leg out of phase with the other Like *During Live Call you here:*>* *>* 1. A: Hi Bro*>* 2. B; Hi Man, How are you??*>* 3. A: Excellent Bro*>* 4. B: Good For you*>* 5. A: Have a nice Day*>* 6. B; Same to you*>* *>* *>* But on the recorder file you hear:*>* *>* 1. A: Hi / B: Hi man How are you??*>* 2. A: Bro*>* 3. B: Good For you*>* 4. B: Same to you / A; Excellent*>* 5. A: Bro*>* 6. A: Have a Nice Day* Not like repeated, buy out of phase ---------------- Are you able to reproduce this on demand? If so how long do you have to record a sound file for it to occur? Is it always the beginning of the sound file that gets copied to the end, or is it X minutes prior to the end of the file? William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518william.king at quentustech.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> On 03/07/2013 08:33 PM, Jesus Ramon Rocha Velazquez wrote: >* Hi,*>* *>* I'm having the next problem:*>* *>* When using record_session on long calls (more than 20 minutes) at some*>* point the final output file "bad mix" the channels, so the A-Leg and*>* B-Leg get one over the other in different times. It looks that in some*>* point the A-Leg "cuts" or "accelerate" audio frames and as result the*>* final audio sounds like two people talking at different times or at the*>* same time but whitout order. Like:*>* *>* During Call Audio:*>* *>* 1. A: Hi Bro*>* 2. B; Hi Man, How are you??*>* 3. A: Excellent Bro*>* 4. B: Good For you*>* 5. A: Have a nice Day*>* 6. B; Same to you*>* *>* *>* On Final Output Audio File:*>* *>* 1. A: Hi / B: Hi man How are you??*>* 2. A: Bro*>* 3. B: Good For you*>* 4. B: Same to you / A; Excellent*>* 5. A: Bro*>* 6. A: Have a Nice Day*>* *>* Tried to use same codec on both legs but not working, the result for*>* long calls is the same. also tried to change the .wav for a .gsm but*>* when converting, the result is the same.*>* *>* Dial Plan:*>* *>* <action application="set" data="continue_on_fail=true"/>*>* <action application="ring_ready" />*>* <action application="set" data="ringback=%(2000,4000,440.0,480.0)"/>*>* <action application="set" data="record_waste_resources=true"/>*>* <action application="set" data="media_bug_answer_req=true"/>*>* <action application="set" data="RECORD_TITLE=Recording*>* ${destination_number} ${caller_id_number} ${strft$*>* <action application="set" data="RECORD_SOFTWARE=OverCall PBX*>* 1.0.0.1"/>*>* <action application="set" data="RECORD_ARTIST=OverCall PBX*>* 1.0.0.1"/>*>* <action application="set" data="RECORD_COMMENT=OverCall PBX*>* 1.0.0.1"/>*>* <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d*>* %H:%M)}"/>*>* <action application="set" data="RECORD_STEREO=true"/>*>* <action application="record_session"*>* data="/my/path/grabaciones/${uuid}.wav"/>*>* *>* <action application="execute_extension" data="SETUP_RECORDING XML*>* default"/>*>* *>* <action application="bridge"*>* data="sofia/gateway/...................*>* *>* I'm Using*>* *>* Intel(R) Core(TM) i5-2310 (4 Cores)*>* 8 GB Ram*>* Debian Wheezy x86*>* FreeSWITCH Version 1.3.14b+git~20130301T214848Z~c35a41e4ca (git c35a41e*>* 2013-03-01 21:48:48Z)*>* *>* Greetings and hope some may know whats happening here, because it's*>* driving me crazy*>* *>* *>* _________________________________________________________________________*>* Professional FreeSWITCH Consulting Services:*>* consulting at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>*>* http://www.freeswitchsolutions.com*>* *>* FreeSWITCH-powered IP PBX: The CudaTel Communication Server*>* http://www.cudatel.com*>* *>* Official FreeSWITCH Sites*>* http://www.freeswitch.org*>* http://wiki.freeswitch.org*>* http://www.cluecon.com*>* *>* FreeSWITCH-users mailing list*>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>*>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users*>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users*>* http://www.freeswitch.org* ------------------------------

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Static linking of libraries into freeswitch
by Siri MM 08 Mar '13

08 Mar '13

Hello, I am working on a system, where I build FS on a development linux ubuntu server, and deploy the built binaries on a target linux ubuntu server (6.06). The problem that I am facing is, the Target server has default libraries installed, and cannot be upgraded to install new libraries (no internet access). When i study the delta between the two servers, missing libraries on the target server are libodbc and liblibjpeg. As a workaround,I would like to bundle these two libraries along with the freeswitch binaries statically. Would appreicaite any inputs on how to go about this. Thanks!

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BUG??? Bad mixed Audio record_session
by Jesus Ramon Rocha Velazquez 08 Mar '13

08 Mar '13

Hi, I'm having the next problem: When using record_session on long calls (more than 20 minutes) at some point the final output file "bad mix" the channels, so the A-Leg and B-Leg get one over the other in different times. It looks that in some point the A-Leg "cuts" or "accelerate" audio frames and as result the final audio sounds like two people talking at different times or at the same time but whitout order. Like: During Call Audio: 1. A: Hi Bro 2. B; Hi Man, How are you?? 3. A: Excellent Bro 4. B: Good For you 5. A: Have a nice Day 6. B; Same to you On Final Output Audio File: 1. A: Hi / B: Hi man How are you?? 2. A: Bro 3. B: Good For you 4. B: Same to you / A; Excellent 5. A: Bro 6. A: Have a Nice Day Tried to use same codec on both legs but not working, the result for long calls is the same. also tried to change the .wav for a .gsm but when converting, the result is the same. Dial Plan: <action application="set" data="continue_on_fail=true"/> <action application="ring_ready" /> <action application="set" data="ringback=%(2000,4000,440.0,480.0)"/> <action application="set" data="record_waste_resources=true"/> <action application="set" data="media_bug_answer_req=true"/> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strft$ <action application="set" data="RECORD_SOFTWARE=OverCall PBX 1.0.0.1"/> <action application="set" data="RECORD_ARTIST=OverCall PBX 1.0.0.1"/> <action application="set" data="RECORD_COMMENT=OverCall PBX 1.0.0.1"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="/my/path/grabaciones/${uuid}.wav"/> <action application="execute_extension" data="SETUP_RECORDING XML default"/> <action application="bridge" data="sofia/gateway/................... I'm Using Intel(R) Core(TM) i5-2310 (4 Cores) 8 GB Ram Debian Wheezy x86 FreeSWITCH Version 1.3.14b+git~20130301T214848Z~c35a41e4ca (git c35a41e 2013-03-01 21:48:48Z) Greetings and hope some may know whats happening here, because it's driving me crazy

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mod_python crashes freeswitch when there's an issue with module, Is this a bug?
by Brian Foster 08 Mar '13

08 Mar '13

I'm not sure if this is a bug or not. I'd think that if there's an issue with a python module, it would just throw an error and keep going. I'm asking to see if this is truly a bug or a feature. Here's the console log: http://pastebin.freeswitch.org/20669 Let me know what you guys think. I can post more information if needed. -- Brian D. Foster Endigo Computer LLC Email: bdfoster(a)endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.

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"make current" has introduced lua dbh errors with ODBC to mysql
by Paul 07 Mar '13

07 Mar '13

I ran a "make current" to our FS installation which was 2 years old, and was working perfectly. After the upgrade we are getting errors with the dbh function, which is called from our lua script, whereby the ODBC driver appears to be working but the submitted SQL commands show an error as if the database is not being accessed. We can run the same commands in MYSQL and get a correct result. I have seen a few threads discussing some changes in the dbh command but cant seem to find a work around. Attached is the LUA code, the "status" command from cli showing version and the error messages when running the test program. Any one have any ideas ?? Thanks in advance We wrote the following test lua script to isolate the problem and have found that the ODBC dri output from status --> FreeSWITCH (Version 1.3.16b git d7a9c18 2013-03-07 00:42:41Z) is ready 2013-03-07 04:17:28.984881 [DEBUG] freeswitch_lua.cpp:352 DBH handle 0xc61fc90 Connected. 2013-03-07 04:17:28.984881 [INFO] switch_cpp.cpp:1274 Successfully connected to the database 2013-03-07 04:17:28.984881 [ERR] switch_core_sqldb.c:1124 SQL ERR: [select count(*) from people] no such table: people 2013-03-07 04:17:28.984881 [ERR] freeswitch_lua.cpp:435 DBH NOT Connected. 2013-03-07 04:17:28.984881 [DEBUG] freeswitch_lua.cpp:370 DBH handle (nil) released. 2013-03-07 04:17:28.984881 [ERR] mod_lua.cpp:198 /usr/local/freeswitch/scripts/db_test.lua:13: assertion failed! LUA code is function debug(s) freeswitch.consoleLog("info", s .. "\n") end -- function debug(s) -- load the freeswitch.Dbh ODBC connector local dbh = assert(freeswitch.Dbh("iamfine")) row = {} if dbh:connected() == true then debug("Successfully connected to the database") local get_size = "select count(*) from people" assert (dbh:query(get_size, function(grow) for key, val in pairs (grow) do row[key] = val testcount = val end end )) debug("The table people IS populated with a record count of :: " ..testcount) end assert(dbh:release()) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/make-current-has-introduced-l… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Freeswitch MWI
by Archana Venugopan 07 Mar '13

07 Mar '13

Hi, When I leave a voicemail MWI is being send to phone. But after listening to my voicemail messages I guess MWI is not being sent to phone, because of which the display in phone is not being refreshed. Can anyone please suggest how am I send MWI after listening to my voicemail. Thanks.

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Test message
by Ken Rice 07 Mar '13

07 Mar '13

Please Ignore Me -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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