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FS Best practices
by Todd 19 Mar '10

19 Mar '10

Hey List, still kinda new to this whole deal, looking for some advice. We are building a website that needs to interact with FS. Not looking for specific instructions just opinions about the best practices for doing things like click to dial from a web page, calling CDR records to web applications to display customers stats and general web integration. Does this all need to be done using scripting languages like Lua or Perl, or can you directly integrate FS with a website in some other manner. Again, just getting into all this, but having a blast learning FS, config and power rock! Anyhow, some places or things I should start looking into would be greatly appreciated. Thanks List!

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make errors - mod_cepstral
by Aloysius Lloyd 19 Mar '10

19 Mar '10

Hi All, I am trying to enable flite & cepstral on a existing FreeSWITCH installation. I follow the following steps 1. build directory cp modules.con.in modules.conf 2. edit modules.conf remove the comments from the following asr_tts/mod_flite asr_tts/mod_cepstral 3../bootstrap.sh 4. ./configure 5. make I get the following error make[5]: *** [mod_cepstral.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_cepstral-all] Error 1 make[2]: *** [all-recursive] Error 1 How can I successfully enable the mod_cepstral and flite. Thanks Lloyd

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Newbie problem with Multiple Tenants and mod_xml_curl
by lloyd thomas 19 Mar '10

19 Mar '10

I eventually managed to get mod_xml_curl working for me. Now I trying to be a bit cleverer setting up a multi-tenancy using mod_xml_curl based on the wiki instrucions http://wiki.freeswitch.org/wiki/Multiple_Companies. But I get an error when a phone tries to register. 2010-03-18 21:19:04.869688 [WARNING] sofia_reg.c:1853 Can't find user [ 203(a)192.168.101.102] You must define a domain called '192.168.101.102' in your directory and add a user with the id="203" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2010-03-18 21:19:04.885327 [WARNING] sofia_reg.c:1018 SIP auth failure (REGISTER) on sofia profile 'internal' for [203(a)192.168.101.102] from ip 192.168.91.28 I think I am nearly there,but I suspect there is something wrong with the resulting 'user directory' xml. --------------------------------------------------------------- <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="User Directory"> <domain name="default"> <groups> <group name="default"> <users> <user id="203"> <params> <param name="password" value="$${default_password}"/> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}\}${sofia_contact(${dialed_user}@${dialed_domain})}"/> <param name="vm-password" value="1234"/> <param name="vm-disk-quota" value="30"/> </params> <variables> <variable name="user_context" value="default"/> <variable name="accountcode" value="203"/> <variable name="day_bar" value="3"/> <variable name="night_bar" value="3"/> <variable name="effective_caller_id_name" value="Extension 203"/> <variable name="effective_caller_id_number" value="203"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_number}"/> </variables> </user> </users> </group> </groups> </domain> </section> </document> -------------------------------------------------------------------- I think the Company Directory XML is OK -------------------------------------------------------------------- <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="Company Directory"> <domain name="default"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <variables> <variable name="record_stereo" value="true"/> <variable name="default_gateway" value="$${default_provider}"/> <variable name="default_areacode" value="$${default_areacode}"/> <variable name="transfer_fallback_extension" value="operator"/> <variable name="user_context" value="default"/> </variables> </domain> <domain name="phisystems.dom"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <variables> <variable name="record_stereo" value="true"/> <variable name="default_gateway" value="$${default_provider}"/> <variable name="default_areacode" value="$${default_areacode}"/> <variable name="transfer_fallback_extension" value="operator"/> <variable name="user_context" value="phisystems.dom"/> </variables> </domain> </section> </document> -------------------------------------------------------------- I get the following result when I 'sofia status' Name Type Data State ================================================================================================= internal profile sip:mod_sofia@192.168.101.102:5060 RUNNING (0) phisystems.dom alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@192.168.101.102:5080 RUNNING (0) ================================================================================================= Thanks in advance

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ProxyMedia Unable to hear voice
by lakshmanan ganapathy 19 Mar '10

19 Mar '10

Hi all, I've updated to the latest SVN 17048. Here is my dialplan: <include> <context name="default"> <extension name="outbound_soc"> <condition field="destination_number" expression="^.*$"> <action application="ring_ready" /> <action application="set" data="continue_on_fail=true"/> <action application="set" data="bypass_media=true"/> <action application="set" data="ignore_early_media=true"/> <action application="set" data="exec_after_bridge_app=park"/> <action application="socket" data=" 0.0.0.0:8447 async full"/> </condition> </extension> </context> </include> >From a nc session I issued the following commands. connect sendmsg call-command: execute execute-app-name: answer sendmsg call-command: execute execute-app-name: bridge execute-app-arg: user/1010 I (user/1001) is able to hear the voice of the bridged party (user/1010). But user/1010 is unable to hear my voice. If I set the bypass_media=false, then both can hear the voice. I wanted to know why when bypass_media=true, the user/1010 cannot hear my voice. Here is the freeswitch log: http://pastebin.freeswitch.org/12484

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Video pass problem
by Mikhail Krivushin 19 Mar '10

19 Mar '10

Hello! We have problem with pass video over FreeSWITCH. I tshark traf, and see that we have 1280 video packets input, and only 560 passed to B leg. Anyone can point me to right direction? I can send pcap file by request. Most time I have black screen, and then one frame can appear, and I see frozen picture pair minutes, and then other frozen picture. We have not network issues, we have good video, when bypass FS. We have not perfomance troubles to. We have ubuntu 9.10 x64, and powerfull server board. uname: Linux fs 2.6.31-17-generic #54-Ubuntu SMP Thu Dec 10 17:01:44 UTC 2009 x86_64 GNU/Linux config: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=BV32,BV16,speex@8000h @20i,speex@16000h@20i,speex@32000h @20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=BV32,BV16,speex@8000h @20i,speex@16000h@20i,speex@32000h @20i,PCMU,PCMA,GSM,H264,H263,H263-1998,H263-2000,H261"/> -- С уважением, Кривушин Михаил г. Томск сот. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME(a)jabber.ru skype: mkrivushin

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Caller ID w/ SLA/SCA
by Gabriel Kuri 19 Mar '10

19 Mar '10

Hey guys, is the Caller ID data (number and name) supposed to be passed between the phones when using SLA/SCA ? If you put a call on hold on phone 1, and pick up the call on phone 2, number/name do not show up. Likewise, if you go back to the same call on phone 1, the name and number are not present. Is this normal? Do I smell a bounty ;) ? Cheers, Gabe

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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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