March 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Missing modules and configs in fresh install
by Free Switcher 30 Mar '15

30 Mar '15

Hello, I'm trying to create a new installation of freeswitch. I'm running CentOS 6.6 and picked the easy path to install pre-built binaries using yum. Looking at /usr/lib64/freeswitch after the install, I noticed that mod_conference.so wasn't there. Also noticed that no config files were installed in /etc/freeswitch. Is this expected? I downloaded configs from the latest stash repository and I was then able to start freeswitch with vanilla config. I see several startup messages about missing mods. Which modules should I expect to see as part of the pre-built binaries? How/where does one download additional mods? Here is what I have in the /usr/lib64/freeswitch : mod_cdr_csv.so mod_commands.so mod_console.so mod_dialplan_directory.so mod_dialplan_xml.so mod_dptools.so mod_event_socket.so mod_logfile.so mod_loopback.so mod_native_file.so mod_sndfile.so mod_sofia.so mod_spandsp.so mod_syslog.so mod_tone_stream.so mod_xml_rpc.so Any help is appreciated. Thanks, Andy

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USB GSM Dongle with FreeSwitch for voice calls
by Luiz Fernando Softov 29 Mar '15

29 Mar '15

Hi ***** - Enable coredump http://blog.urdada.net/2007/12/31/71/ ***** - Find it after next crash ***** - Analyze coredump First, when i say crash, it's a panic in USB, not a crach of daemon/executable. This is because Freeswitch/GsmOpen is attached in TTY and lock the TTY, blocking other processes to manage USB (tty). ***** Find what is calling USB modem power off and disable it.***** No poweroff, no error. Some modems power off and power on again, i try to find what is causing this error, without success. Maybe some errors in power suply or modem just reset, because i try it with 4, 10, 20 and 30 modems, and the problem persists. In my search, google and other forums, i found someone talking about modem resetting for own. Sometimes the user wan't to change de SIM in modem, and remove it and put it again, without make a dettach in my interface/system. Or if he not remove the modem and just change de SIM (o.O), its a request of many users of my system using E303 or other modems. I identify this, because i'm attached in ttyU0.0, ang get event im my source or a CME-ERROR 10, if modem just changed SIM, then i reset the modem with AT command ^RESET. I'm using mod_xml_curl to get interfaces info, and ESL events to manage this... Then i make a "gsm remove xxxx" and when modem arrive i make a "gsm reload". This way not affect the calls in progress. I talk with Shlomi Agiv, and apply some functions of his patch, now i'm just making tests.... Now i'm having some problems, just when i call a "gsm remove" in ESL, its stop to reply, even when i send other command, like "sofia status". It's the same when i send command with fs_cli... Before ask, i make some debugs, change source, put log and other things.. I ask, because some times other people have the same problem! Thanks for the reply... -- Luiz Fernando Softov http://www.softov.com.br fernando(a)softov.com.br

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Re: [Freeswitch-users] Transfering Attended Call
by Luiz Fernando Softov 29 Mar '15

29 Mar '15

Hi, i'm executing "att_xfer" https://freeswitch.org/confluence/display/FREESWITCH/Attended+Transfer So... A Bridge to B, and i make a att_xfer using B uuid, to C... When B in call with C. A gets silence, i try to play a music with playback, and its work, but i can't execute anything until music is played. And when B digits *, to start with transfer, i try to play a music to A uuid, but i can't receive other DTMF events from B... Is there a way to play a music to A, when i make a att_xfer? Then when att_xfer is done, de music stop to play.... I try to use "fifo" too, but is the same result, i can't get B leg DTMF events after... i'm not using lock in this Events... I'm just want to play some music or sound to A, until he is waiting for B or C to take the call -- Luiz Fernando Softov http://www.softov.com.br fernando(a)softov.com.br

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Re-establish connection within a SIP session
by Mateus Dalepiane 28 Mar '15

28 Mar '15

We have the following scenario: The session is established between WebRTC and FreeSWITCH using Websockets. Once the session is established, if the websocket connection drops the media continues to flow util FreeSWITCH tries to send a re-INVITE to the client. At this point it realizes that the connection was closed and hangs up the call. Now, if the websocket connection drops and is re-established, would it be possible to inform FreeSWITCH that the new connection should be used for the previously established session? If the WebRTC client sends an INVITE message with the old session parameters, FreeSWITCH will be able to understand that it belongs to the old session?

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USB GSM Dongle with FreeSwitch for voice calls
by Luiz Fernando Softov 28 Mar '15

28 Mar '15

I thy to open pastebin https://pastebin.freeswitch.org/23657, but it's without content. I have a lot of problem with modem/dongle Lets say i use /dev/ttyU3.2 to audio device /dev/ttyU3.3 to control device Some times the modem power off and crash my usb controller I tried using different power supplies 40A, 60A, but no sollution I use FreeBSD, and when some dongle reset usbconfig stop reply, and i need to restart the machine When its crash, i can see this ttys remaining /dev/ttyU3.3 /dev/ttyU3.3.init /dev/ttyU3.3.lock The others tty close normally, i think this crash because freewitch still attached in ttyU3.3 Someone can help me? -- Luiz Fernando Softov http://www.softov.com.br fernando(a)softov.com.br

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Strange effect with delay_echo against VAD
by Stanislav Sinyagin 28 Mar '15

28 Mar '15

I got a strange effect, and help in understanding it would be appreciated. My FreeSWITCH server registers at an ITSP, and forwards incoming calls to "echo" or "delay_echo" applications, immediately after answering. The ITSP uses VAD and comfort noise when it sends audio to my server. If the call is sent to "echo" application, the calling party hears the echo, and also it hears comfort noise. If the call is sent to "delay_echo", with any delay (I tried 200 to 1000ms), the calling party hears only a loud noise. Here's a recording on the calling side (my voice in the left ear) http://www.k-open.com/s/0202eb6bfa655e47974325574ac47c7b.wav Also if the call is sent to an IVR, and delay_echo is executed as an IVR action, the caller hears the echo as expected. The call is happening in PCMA. I've got also packet captures, will share them on request, as they contain some confidential information.

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transfer_fallback_extension
by T Fred Farmington 27 Mar '15

27 Mar '15

On a number of websites and in my conf/directory/default.xml file I see: <variable name = "transfer_fallback_extension" value = "operator"/> And in my conf/dialplan/default.xml file I see <extension name = "operator"> where the operator 'extension' is defined as a single extension number. But I don't want the failed transfer attempt to go to a single pre-defined extension I want the failed transfer attempt to go back to the extension which initially launched the transfer attempt How/where would I modify the variable transfer_fallback_extension to get this to work as needed? Or should this be handled in a different manner somewhere else within FreeSWITCH? If so, where/how? Thanks

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 27 Mar '15

27 Mar '15

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 FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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bind_meta_app - bind to * AND #
by Денис 27 Mar '15

27 Mar '15

Hello. I want use bind_meta_app, how i can bind on *2 and #1? If i set bind_meta_key=# - bind on 2 BEFORE this set - bound to #2. I don`t want use bind_digit_action - it's so strange and incomprehensible, especially bind to leg B My last try: <action application="bind_digit_action" data="att_xfer,*2,exec:execute_extension,att_xfer XML features"/> <action application="bind_meta_app" data="2 b b execute_extension::att_xfer XML features"/> <action application="set" data="bind_meta_key=#"/> <action application="bind_digit_action" data="blind_xfer,#1,exec:execute_extension,dx2 XML features"/> <action application="bind_meta_app" data="1 b b execute_extension::dx XML features"/> Who have working configs for transfer (blind+att) with A+B legs?

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Swap on att_xfer application
by Jayesh Nambiar 27 Mar '15

27 Mar '15

Hello, I was exploring the usage of att_xfer and it works as expected. I was just wondering that is there a way when the last party answers, I can hold that party and get back to initial call placing the last party on hold using some DTMF action?? I see a DTMF action like # to end the last call and get back to the initial call, but what I am looking at is hold the last call and get back to initial call before I actually transfer the call. Is there anything similar available which can be used to swap between calls using the att_xfer application. Thanks, --- Jayesh

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