March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How do I use eavesdrop?
by Mark Sobkow 18 Mar '10

18 Mar '10

I know eavesdrop is an application, not an API, so you have to use it differently. I have an operator dialing in on extension 5000 being bridged to a client, and I'm getting those lovely little echoes that confirm the call is bridged (both operator and client softphones are sharing my one headset and mic.) In the dialplan, I parse out the extension when someone dials in to 665000 (last 4 digits are the extension to monitor), look up the appropriate operator UUID in my mnesia/Erlang tables, receive the "call" callback from Freeswitch, and build a proplists replacement with the execute-app-name set to "eavesdrop" and execute-app-arg set to the operator UUID. I issue a sendmsg() command to Freeswitch, and it returns ok. However, what happens to the extension that tried to eavesdrop is it gets two events: CHANNEL_PARK and SEND_MESSAGE. The person making the eavesdrop call can't actually hear anything. I've attached the Erlang callback module that the dialplan executes. The dialplan fragment is as follows: <extension name=\"eavesdrop\"> <condition field=\"destination_number\" expression=\"^66(\\d\\d\\d\\d)$\"> <action application=\"answer\"/> <action application=\"set\" data=\"eavesdrop_extension=$1\" /> <action application=\"erlang\" data=\"pbx_eavesdrop pursuit@testsrv\" /> </condition> </extension> Unfortunately I can't think of any way to look up the UUID other than using the mnesia tables from the application. When I tried the sample dialplans from Freeswitch, the eavesdrop module itself _did_ work. I'm just having trouble cobbling together the "glue" for Erlang. P.S. This is the last Freeswitch feature I have to code for this release, other than niggling cleanups like adding a few more voice prompts, which is easy enough as I've already sample code on how to do it. With a little luck we'll make our April 1st ready-for-testing deadline! We wouldn't have made it without Andrew's hints, tips, and tricks for using Erlang with Freeswitch. Thanks a lot! :D -- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow(a)marketelsystems.com Web: http://www.marketelsystems.com %% Author: mark %% %% Created: Mar 8, 2010 %% %% Description: %% This callback is executed when someone dials in to an eavesdrop %% -module(pbx_eavesdrop). %% %% Exported Functions %% -export([start/0, run/0, launch/1]). %% %% Include files %% -include_lib("figure/include/gut.hrl"). -include_lib("pbx/include/pbx_mnesia.hrl"). start() -> Pid = spawn( ?MODULE, run, [] ), register( ?MODULE, Pid ), syslog:debug( "pbx_eavesdrop:start() Started eavesdrop thread ~p~n", [Pid] ), { ok, Pid }. run() -> receive { call, Data } -> { event, [UUID | Rest]} = Data, EavesdropExtension = proplists:get_value( "variable_eavesdrop_extension", Rest ), %% syslog:debug( "pbx_eavesdrop:run() Rest=~p~n", [Rest] ), syslog:debug( "pbx_eavesdrop:run() New eavesdrop received for extension=~p~n", [EavesdropExtension] ), case pbx_mnesia:lookupOperatorByExtension( EavesdropExtension ) of { ok, ExtensionRegistry } -> case proplists:is_defined( "execute-app-name", Rest ) of true -> PropsNoAppName = proplists:delete( "execute-app-name", Rest ); _ -> PropsNoAppName = Rest end, case proplists:is_defined( "execute-app-arg", PropsNoAppName ) of true -> PropsNoAppArg = proplists:delete( "execute-app-arg", PropsNoAppName ); _ -> PropsNoAppArg = PropsNoAppName end, PropsWithAppName = PropsNoAppArg ++ [ proplists:property( "execute-app-name", "eavesdrop" ) ], PropsWithAppArg = PropsWithAppName ++ [ proplists:property( "execute-app-arg", ExtensionRegistry#pbx_operator_registry.operator_uuid ) ], %% syslog:debug( "pbx_eavesdrop:run() sendmsg( ~p, ~p )~n", [ UUID, PropsWithAppArg ] ), EavesdropResult = pbx:sendmsg( UUID, PropsWithAppArg ), syslog:debug( "pbx_eavesdrop:run() EavesdropResult = ~p~n", [EavesdropResult] ); Error -> %% Operator not logged in to extension syslog:info( "pbx_eavesdrop:run() Could not locate extension ~s, Error ~p~n", [ EavesdropExtension, Error ] ), _BgapiResult = pbx:bgapi( uuid_broadcast, UUID ++ " " ++ "/opt/freeswitch/sounds/en/us/callie/misc/8000/sorry.wav both" ), _ResultHangup = pbx:bgapi( sched_hangup, "+3 " ++ UUID ) end, run(); {call_event, Data} -> { event, [UUID | Rest]} = Data, EventName = proplists:get_value( "Event-Name", Rest ), syslog:debug( "pbx_eavesdrop:run() call_event UUID=~p, EventName=~p~n", [UUID, EventName] ), run(); {get_pid, _UUID, Ref, Pid} -> NewPid = spawn( ?MODULE, run, [] ), syslog:debug( "pbx_eavesdrop:run() Request to spawn new handler process, returning PID ~p~n", [NewPid] ), Pid ! { Ref, NewPid }, run() end. launch( Ref ) -> NewPid = spawn( ?MODULE, run, [] ), syslog:debug( "pbx_eavesdrop:launch() Returning new PID ~p~n", [NewPid] ), {Ref, NewPid}.

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PaeTec
by Ken Fulmer 18 Mar '10

18 Mar '10

Has anyone created a gateway to connect to PaeTec? Thanks, Ken Fulmer

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moh background in Lua
by Tjardick van der Kraan 18 Mar '10

18 Mar '10

Hi. I'm trying to create a lua script but would like a certain task (music on hold) to be 'backgrounded': -- Loop while the call is active while (session:ready() == true) do -- do something here session:execute("playback", moh) freeswitch.consoleLog("INFO","Started music on hold.\n"); -- do some stuff while playing music session:sleep(30000); -- Maybe we need a way to stop music or playback will stop automatically? session:execute("playback", "/sounds/we_will_answer_asap.wav") freeswitch.consoleLog("INFO","Play'ed announcement.\n"); end "ofcourse" currently the only thin that happens is the music and my loop does not continue. Is there a way to start/stop music in the background? I guess in Asterisk language i will need the command StartMusicOnHold... Thanks for any suggestions. Regards, Tj

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Disable call waiting
by Adam Ford 18 Mar '10

18 Mar '10

I apologize if this has been answered before, but I can't seem to find anything on google. Is there a way to disable call waiting in FreeSWITCH? -Adam

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intercept in fifo queue
by Vladimir Elizarov 18 Mar '10

18 Mar '10

Hello. How can I intercept a fifo? I tried to use the example of intecept default dialplan, but in no fifo dialed_extension. <extension name="operators"> <condition field="destination_number" expression="^operators$"> <action application="bind_meta_app" data="2 b s record_session::/opt/freeswitch/storage/record/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="answer"/> <action application="playback" data="ivr/ivrvoice/razgovor-mozhet-byt-zapisan.wav"/> <action application="playback" data="ivr/ivr-hold_connect_call.wav"/> <action application="set" data="fifo_music=$${hold_music}"/> <action application="fifo" data="operators@$${domain} in undef"/> </condition> </extension> <fifo name="operators@$${domain}" importance="0"> <member timeout="15" simo="1" lag="20">{fifo_caller_exit_key=9,fifo_consumer_exit_key=9}user/100@$${domain}</member> </fifo> </fifos> -- Best regards, Vladimir Elizarov

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mod_fifo recording with fifo_record_template
by Steven Ward 18 Mar '10

18 Mar '10

I'm looking to use fifo_record_template to specify uniquely named recording files. Here is the situation: I'm using a fifo with on-hook members. I would like each fifo call to be recorded, with each recording-file named uniquely. Thanks to some helpful IRC talk, I have learned that the following works well: <fifo name="Outside_Call@$${domain}" importance="0"> <member timeout="20" simo="1" lag="0">{fifo_record_template=$${base_dir}/recordings/my_recording.wav,fifo_member_wait=nowait}user/35351@ $${domain}</member> </fifo> However, this records each and every fifo call to the file (in my case): /usr/local/freeswitch/recordings/my_recording.wav Every new call overwrites the file. Instead of a statically defined file name, I would like a unique file name for every fifo call. I don't know how to make reference to channel variables in this definition. I tried the following, but it results in an error: <fifo name="Outside_Call@$${domain}" importance="0"> <member timeout="20" simo="1" lag="0">{fifo_record_template=$${base_dir}/recordings/{${uuid}.wav,fifo_member_wait=nowait}user/35351@ $${domain}</member> </fifo> The error I see on the CLI when a call goes into the FIFO is this: 2010-03-18 13:51:11.289448 [CRIT] switch_channel.c:759 Invalid data (${fifo_record_template} contains a variable) It was also suggested to me that I escape the '$' character on the variable. Like this: <fifo name="Outside_Call@$${domain}" importance="0"> <member timeout="20" simo="1" lag="0">{fifo_record_template=$${base_dir}/recordings/{\${uuid}.wav,fifo_member_wait=nowait}user/35351@ $${domain}</member> </fifo> This also resulted in: 2010-03-18 13:52:37.014742 [CRIT] switch_channel.c:759 Invalid data (${fifo_record_template} contains a variable) The recording file doesn't get generated. Any recommendations on using fifo_record_template to generate recording file names that are unique to each call? Is there a way to get channel variables to be evaluated and populate the value of fifo_record_template? Or does fifo_record_template only support static values (meaning I'd have to do a record_session in dial plan?)? Thanks for any feedback.

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Calls
by Jonas Gauffin 18 Mar '10

18 Mar '10

Hello, Is it possible to list calls that are terminated in IVR's, are parked or in early state? "show calls" api just lists established calls, correct? Best regards, Jonas

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Re: [Freeswitch-users] voicemail options [temporarly solved]
by Madovsky 18 Mar '10

18 Mar '10

Finally I succeed to send voicemail with sendmail through a little php script (could be Perl also) it seems that sendmail 8.14.3 has a bug. But call it from PHP the bug doesn't exist. I attached my php script and switch_utils.c. Not sure if the FS developers will accept my modification Regards Franck ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 16, 2010 4:35 PM Subject: Re: [Freeswitch-users] voicemail options ok, I tried 50 times to leave a message, it worked only one time. It seems that sendmail needs to be free to send the email.... ----- Original Message ----- From: devel(a)thom.fr.eu.org To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 16, 2010 3:59 PM Subject: Re: [Freeswitch-users] voicemail options Are you running debian, I have the same problem on debian x64 (already mentioned here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024733.…) and cannot fix it. I don't know if it's a priviledge problem, as I got the same results running FS with or without -u. François De : freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Anthony Minessale Envoyé : mardi 16 mars 2010 20:30 À : freeswitch-users(a)lists.freeswitch.org Objet : Re: [Freeswitch-users] voicemail options are you running FS as root or a regular user? maybe there is a permission or resource limitation problem preventing it from getting the larger stack space necessary? On Tue, Mar 16, 2010 at 2:18 PM, Madovsky <infos(a)madovsky.org> wrote: I noticed that sendmail makes a segfault when FS is trying to send email and sendmail has some emails locked in folder queue.... Regards Franck _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 ---------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org 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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Calculate the Call Duration ( BillSec)
by Aloysius Lloyd 18 Mar '10

18 Mar '10

Hi All, The following lua script for calculate the call duration. But the issue here , the duration is calculating from the outbound initiate time. How to get the call answer time? session:preAnswer(); -- sleep a second session:sleep(1000); -- digits = session:playAndGetDigits(10, 20, 3, 5000, "#", "enter-dest.wav", "invalid-digits.wav", "\\d+|\\*"); -- Initiate an outbound call outSession = freeswitch.Session("sofia/gateway/voipms/"..digits,session) starttime = os.time(); freeswitch.consoleLog("info", "Start Time"..starttime.." \n"); freeswitch.consoleLog("info", "Before Bridge the Call \n"); freeswitch.bridge(session, outSession); freeswitch.consoleLog("info", "After Bridge the Call \n"); endtime = os.time(); freeswitch.consoleLog("info", "End Time"..endtime.." \n"); billsec = os.difftime(endtime,starttime); freeswitch.consoleLog("info", "Blling Sec"..billsec.." \n"); -- hangup session:hangup(); Thanks Lloyd

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Newbie question on domain name
by false 18 Mar '10

18 Mar '10

Hi List i feel strange about default domain in Freswitch: i read the vars.xml and default domain name is IP address: <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> and i register successfull the 1000, 1002 extension the i copy 1000.xml to 7000.xml then i register 7000 extension but get error Please help Ha` Registrations: ================================================================================================= Call-ID: 56dc9e359865070c(a)192.168.1.6 User: 1002(a)192.168.1.11 Contact: "user" <sip:1002@192.168.1.6:4318;user=phone> Agent: Grandstream BT110 1.0.7.3 Status: Registered(UDP)(unknown) EXP(2010-03-18 04:46:13) Host: fs IP: 192.168.1.6 Port: 4318 Auth-User: 1002 Auth-Realm: 192.168.1.11 MWI-Account: 1002(a)192.168.1.11 Call-ID: 8fa759179877592f(a)192.168.1.12 User: 1009(a)192.168.1.11 Contact: "user" <sip:1009@192.168.1.12:57386;user=phone> Agent: Grandstream BT110 1.0.7.3 Status: Registered(UDP)(unknown) EXP(2010-03-18 04:48:08) Host: fs IP: 192.168.1.12 Port: 57386 Auth-User: 1009 Auth-Realm: 192.168.1.11 MWI-Account: 1009(a)192.168.1.11 ================================================================================================= freeswitch@fs> 2010-03-18 02:52:00.391683 [WARNING] sofia_reg.c:1866 Can't find user [7000(a)192.168.1.11] You must define a domain called '192.168.1.11' in your directory and add a user with the id="7000" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2010-03-18 02:52:00.423766 [WARNING] sofia_reg.c:1024 SIP auth failure (REGISTER) on sofia profile 'internal' for [7000(a)192.168.1.11] from ip 192.168.1.2 Thiết kế ngay một Pingbox độc đáo cho riêng bạn! Tạo một nơi để chat trên blog là chuyện nhỏ. http://vn.messenger.yahoo.com/pingbox/

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