August 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem receiving fax
by Antonio 07 Sep '11

07 Sep '11

Hi to all, I'm having problems receiving fax in a pri E1 line. When a restart freeswitch or the server and i can receive one fax and then it starts to fail again. I didn't open a jira bug, because I'm not sure where to look for the problem. The log can be found at http://pastebin.freeswitch.org/17047 Thanks, António Silva

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DTMF from Nortel BCM 400 to Freeswitch
by Ray Pang 07 Sep '11

07 Sep '11

I’ve been unable to get DTMF to work from BCM 400 to Freeswitch PSTN -> BCM -> sip forward -> Freeswitch Autoattendant. DTMF never gets recognized: 2011-08-30 13:13:46.164796 [DEBUG] switch_ivr_menu.c:428 Executing IVR menu auto_attendant_1 2011-08-30 13:13:46.164796 [DEBUG] switch_ivr_play_say.c:1278 Codec Activated L16@8000hz 1 channels 30ms 2011-08-30 13:13:46.174819 [DEBUG] sofia.c:4761 Channel sofia/sipinterface_1/3335 entering state [ready][200] 2011-08-30 13:13:46.412814 [DEBUG] switch_rtp.c:3083 Correct ip/port confirmed. 2011-08-30 13:13:46.940791 [DEBUG] sofia.c:6308 dispatched freeswitch event for INFO 2011-08-30 13:13:47.360757 [DEBUG] sofia.c:6308 dispatched freeswitch event for INFO 2011-08-30 13:13:48.350702 [DEBUG] sofia.c:6308 dispatched freeswitch event for INFO 2011-08-30 13:13:48.390701 [DEBUG] sofia.c:6308 dispatched freeswitch event for INFO 2011-08-30 13:13:48.397689 [DEBUG] sofia.c:6308 dispatched freeswitch event for INFO 2011-08-30 13:13:48.872658 [DEBUG] switch_ivr_play_say.c:1648 done playing file 2011-08-30 13:13:48.872658 [DEBUG] switch_ivr_menu.c:343 waiting for 4/4 digits t/o 2000 2011-08-30 13:13:58.893140 [DEBUG] switch_ivr_menu.c:390 digits '' Packet capture shows this tidbit: p=Digit-Collection y=Digits d=7 I’m very new to this, any help or advice will be appreiciated. TIA, RP

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Can't make mod_rtmp
by peely 07 Sep '11

07 Sep '11

Hi, I'm having trouble building FreeSWITCH with the latest mod_rtmp goodness. On Ubuntu X64 compiling with gcc I get: cc1: warnings being treated as errors /usr/src/freeswitch/src/mod/endpoints/mod_rtmp/mod_rtmp.c: In function ‘rtmp_function’: /usr/src/freeswitch/src/mod/endpoints/mod_rtmp/mod_rtmp.c:1612:114: error: comparison between ‘switch_call_cause_t’ and ‘enum <anonymous>’ I'm not sure if I need to do anything else, I just naively added endpoints/mod_rtmp to modules.conf and did a make clean install? Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Variables in Dialplan - Problem with getting Variables from User Directory
by Silvio Escher 06 Sep '11

06 Sep '11

Hi there, iam really getting crazy these days ;-) I've changed our Freeswitch Version some Weeks ago ( cannot remember exactly but proably from 1.0.6 ) to the git Head. ). Since this Change ( or better since the followed Config File Adaptions ;-) ) i noticed that i cannot use User Variables/Params inside the XML-Dialplan anymore. Some (relevant?! ;) ) Copy&Paste .. my Userentry <include> <user id="99"> <params> <param name="password" value="something"/> <param name="vm-mailto" value="maybe_me(a)omeco.de"/> <param name="vm-mailfrom" value="noreply(a)omeco.de"/> <param name="vm-enabled" value="true"/> <param name="vm-password" value="user-choose"/> <param name="vm-email-all-messages" value="false"/> <param name="vm-notify-email-all-messages" value="true"/> <param name="vm-keep-local-after-email" value="true"/> </params> <variables> <variable name="outbound_caller_id_name" value="Omeco GmbH"/> <variable name="outbound_caller_id_number" value="06316803540"/> <variable name="effective_caller_id_name" value="Silvio Escher"/> <variable name="effective_caller_id_number" value="40"/> <variable name="callgroup" value="1"/> </variables> </user> </include> some dp snippet.. before the change something like <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> was working fine .. Actually iam using <action application="set" data="effective_caller_id_name=${user_data(${username}@${domain_name} var outbound_caller_id_name)}"/> <action application="set" data="effective_caller_id_number=${user_data(${username}@${domain_name} var outbound_caller_id_number)}"/> or an <action application="set_user" data="$1@${domain_name}"/> to get some or all Variables. But i Noticed that the "set_user" Thingie just gets me the Variables - not the Params. So i've still Issues with mod_voicemail ( Mailto is undef ) 2011-08-30 22:02:32.904211 [DEBUG] switch_utils.c:709 Emailed data to [(null)] 2011-08-30 22:02:32.904211 [DEBUG] mod_voicemail.c:2809 Sending notify message to (null) Maybe i has something todo with an missing auth or wrong Domain or whatever .. but i have no clue where to look further. Any Help ( from little Hints till complete Miracle Solve ;-) ) is actually very welcome. Thanks in Advance, Silvio -- Silvio Escher omeco GmbH

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Multiple Tone Detect
by Rob Hutton 06 Sep '11

06 Sep '11

I am trying to get tone detection set up so that a single number can be used for both voice and fax. Tone detection needs to run concurrent to the normal call processing, so that a call proceeds normally unless the fax tones are heard. On this wiki page (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect) it gives the format: <action application="tone_detect" data="fax 1100 r +5000 transfer fax XML default" /> and an associated dialplan entry for the call to be transfered to that actually handles the fax reception. On this wiki page (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect) it says that the format from the first page is not the preferred method, and that the format should be something like: <action application="set" data="execute_on_tone_detect=transfer fax XML default"/> <action application="set" data="tone_detect_hits=1"/> <action application="set" data="execute_on_media=tone_detect mytone 1100 r +5000"/> My question is, using the first format, I could set up multiple detections, eg. one for fax tones that sends/receives a fax, one for answering machines tone that hangs up, etc. Is this possible with the preferred syntax?

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session:recordFile(-) always creates empty wav file during outbound IVR call
by king2kin 02 Sep '11

02 Sep '11

Hi folks, With Lua script and/or originate command, I have tried recording a message file during outbound IVR call over and over, session:recordFile(-) inside Lua script does create a wav file during each of my testings but the recorded audio file is always empty. However, session:recordFile(-) works well for inbound IVR call. I tried the session:recordFile(-) via Lua script in three ways: 1. run lua script "test_outcall_ivr.lua" at freeswitch command-line: luarun test_outcall_ivr.lua -- [test_outcall_ivr.lua] { local sessionx = freeswitch.Session("sofia/gateway/sip.tpad.com/1726011", session) -- Set the path separator pathsep = '/' -- Windows users do this instead: -- pathsep = '\' -- Answer the call -- sessionx:answer() --Create a string with path and filename of a sound file prompt = "ivr" .. pathsep .. "ivr-welcome_to_freeswitch.wav" -- Print a log message freeswitch.consoleLog("INFO","Prompt file is '" .. prompt .. "'\n") --Play the prompt sessionx:streamFile(prompt) -- Record record file sessionx:streamFile("phrase:voicemail_record_message") -- Play a ""bong"" tone prior to recording sessionx:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000, 0, 640)") -- record a message filename = sessionx:getVariable('sounds_dir') .. pathsep .. "123.wav" sessionx:recordFile(filename,300,100,10) -- play back the recorded msg sessionx:streamFile(filename) -- Hangup sessionx:hangup() } 2. I also tried it differently by submitting the following commands at the FreeSwitch command-line interface: originate user/1005 &transfer(8887 xml default) originate user/1005 &lua('test1.lua') originate sofia/gateway/sip.tpad.com/1726011 &lua('test1.lua') -- [test1.lua] { -- Set the path separator pathsep = '/' -- Windows users do this instead: -- pathsep = '\' --Answer the call session:answer() --Create a string with path and filename of a sound file prompt = "ivr" .. pathsep .. "ivr-welcome_to_freeswitch.wav" -- Print a log message freeswitch.consoleLog("INFO","Prompt file is '" .. prompt .. "'\n") --Play the prompt session:streamFile(prompt) -- Record record file session:streamFile("phrase:voicemail_record_message") -- Play a ""bong"" tone prior to recording session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000, 0, 640)") -- record a message filename = session:getVariable('sounds_dir') .. pathsep .. "123.wav" session:recordFile(filename,300,100,10) -- play back the recorded msg session:streamFile(filename) -- Hangup session:hangup() } -- [xml dialplan for extension 8887]: { <extension name="Simple Lua Test"> <condition field="destination_number" expression="^(8887)$"> <action application="set" data="record_waste_resources=true"/> <action application="lua" data="test1.lua"/> </condition> </extension> } ====================================================== For all the above testing cases, session:recordFile(-) always creates an empty wav file for each of outbound IVR calls, however, if I make an inbound IVR　call to run Lua script "test1.lua", session:recordFile(-) always works perfect to generate a normal wav file. So, what's wrong with [session:recordFile(-)] during an outbound IVR call? x.k.

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Start FreeSWITCH as a Windows service with -nonat option
by Aleš Zelenik 01 Sep '11

01 Sep '11

Hello all, I am having trouble starting fs as a windows service with -nonat option If I run from command prompt "FreeSwitchConsole -nonat" it works ok, but I am left with open window in logged-in session. Command to run as a service is "...path...\FreeSwitchConsole.exe -service FreeSWITCH", which can be also verified with "sc qc FreeSWITCH" If I want to add an argument -nonat with command: sc config FreeSWITCH binPath= "C:\Program Files (x86)\FreeSWITCH \FreeSwitchConsole.exe -nonat -service FreeSWITCH" service won't start anymore also, fs cannot be installed as a service with optional arguments, eg FreeSwitchConsole -install -nonat Anyone managed to run fs like this? Thanks, -- Ales Zelenik,

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Sangoma is hiring!
by Moises Silva 31 Aug '11

31 Aug '11

Hello everyone, I apologize in advance for the spam. Sangoma is looking for talented software engineers for development and QA. Roughly speaking we would love to have people with open source mentality and that knows his/her way around Linux and multithreaded C. Kernel level coding is an asset. They should be willing to relocate to work in our offices at Toronto, Canada. Full description in LinkedIn: http://ca.linkedin.com/jobs/jobs-Software-Engineer-VoIPTDMVideoNetworking-1… Interested candidates can contact me directly. Moises Silva Senior Software Engineer, Software Development Manager Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy(a)sangoma.com

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OS X PPC Leopard error on "make"
by Peter Schrock 31 Aug '11

31 Aug '11

I am new to FreeSWITCH (sort of, tried it a while back and dropped it). I have a powerpc Apple running Leopard on it. I followed the directions by Mario G and managed to get all the way up to where he discusses error's with FLITE. I managed to go without the error message because I followed the instructions on how to fix it. However, I have come up with this error message and can't seem to make any sense of it. Help Please. Here is the error I get: make[8]: *** No rule to make target `tport/libtport.la', needed by ` libsofia-sip-ua.la'. Stop. make[7]: *** [all-recursive] Error 1 Making all in packages make[6]: *** [all-recursive] Error 1 make[5]: *** [all] Error 2 make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/ libsofia-sip-ua.la] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Peter

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Condition based on custom sip header
by Alex Massover 31 Aug '11

31 Aug '11

Hi, I'm trying to implement dialplan condition based on custom SIP header ("Header1: value1"). As far as I understood there's no way to read custom SIP header which isn't X- header with sip_h_. Is that correct? What are the other options, please? Is lua's getHeader() suitable for this? -- Best Regards, Alex Massover This mail was sent via Mail-SeCure System.

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