December 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Lua not playing any WAV files
by Marc de Corny 02 Dec '10

02 Dec '10

Hi all, I have run into an issue on something so basic that I must be as simple as enabling a feature somewhere. I have been trying to get lua to play a message from a WAV file. I have tried session:execute("playback", main_msg) and session:streamFile(ivr_invalid_msg) but neither of them play any music to the caller. I tried both to answer and preAnswer the call first but it made no difference. However if I put the same file into the XML dialplan and play it with the commands below I hear the music fine. <action application="set" data="playback_terminators=#"/> <action application="playback" data="${401_sound_file_path}"/> The issue only seems to be from lua when playing any type of wav file and those files are definitelly there as can be read by the XML The error message is below for the execute(playback) command, but nothing can be seen for the 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application playback Requires media! pre_answering channel sofia/external/2031701665@194.0.147.16:5060 EXECUTE sofia/external/2031701665@194.0.147.16:5060playback(/usr/local/freeswitch/sounds/svc_sound_files/default_autoattendant.wav) But there is no mention of the streamFile command. I have had similar issue with the PlayAndGetDigits command. Is there something that I need to enable in lua so that is can playback messages to the caller. Many thanks to anyone who can help. Marc below is the XML dialplan and lua script as well as the log at the very end. XML DIALPLAN: <include> <extension name="IVR_FROM_MYSQL"> <condition field="caller_id_number" expression="Anonymous" break="never"> <action application="set" data="effective_caller_id_number=0000000000"/> </condition> <condition field="destination_number" expression="^(404)"> <action inline="true" application="lua" data="ivr_mysql.lua ${destination_number:3}"/> <action application="set" data="effective_caller_id_name=${404_tag}"/> <action application="bridge" data="${404_dial}"/> </condition> </extension> </include> The LUA script ivr_mysql.lua is callsed and this is it. -- IVR : PLAY IVR WAV FILES -- Global Variables: local dialstr_prefix = "sofia/gateway/CS2k/" local dialstr_main = "" local breakoutcode = "184" local sound_file_folder = "/usr/local/freeswitch/sounds/svc_sound_files/" local ddi = argv[1] -- answer the call session:preAnswer(); freeswitch.consoleLog("info", "All Answered\n"); ivr_invalid_msg = sound_file_folder .. "invalid_msg.wav" main_msg = sound_file_folder .. "default_autoattendant.wav" -- Play with Execute session:execute("playback", main_msg) -- Play with StreamFile session:streamFile(ivr_invalid_msg); dialstr_main = dialstr_main .. dialstr_prefix .. breakoutcode .. "02031701665" session:setVariable("404_dial",dialstr_main) session:setVariable("404_tag","IVR") RELEVANT LOGS : Dialplan: sofia/external/2031701665@194.0.147.16:5060 Regex (PASS) [IVR_FROM_MYS QL] destination_number(4042031956241) =~ /^(404)/ break=on-false Dialplan: sofia/external/2031701665@194.0.147.16:5060 Action lua(ivr_mysql.lua $ {destination_number:3}) INLINE EXECUTE sofia/external/2031701665@194.0.147.16:5060 lua(ivr_mysql.lua 2031956241 ) 2010-12-01 11:56:01.525426 [INFO] switch_cpp.cpp:584 Sending early media 2010-12-01 11:56:01.525426 [DEBUG] sofia_glue.c:2972 AUDIO RTP [sofia/external/2 031701665@194.0.147.16:5060] 10.5.2.105 port 29900 -> 194.0.147.164 port 50202 c odec: 8 ms: 20 2010-12-01 11:56:01.525426 [DEBUG] switch_rtp.c:1418 Starting timer [soft] 160 b ytes per 20ms 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3190 Set 2833 dtmf send payload to 101 2010-12-01 11:56:01.532280 [DEBUG] sofia_glue.c:3195 Set 2833 dtmf receive paylo ad to 101 2010-12-01 11:56:01.532280 [DEBUG] mod_sofia.c:2172 Ring SDP: v=0 o=FreeSWITCH 1291174661 1291174662 IN IP4 10.5.2.105 s=FreeSWITCH c=IN IP4 10.5.2.105 t=0 0 m=audio 29900 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2010-12-01 11:56:01.532280 [NOTICE] mod_sofia.c:2175 Pre-Answer sofia/external/2 031701665@194.0.147.16:5060! 2010-12-01 11:56:01.532280 [DEBUG] switch_channel.c:2544 (sofia/external/2031701 665@194.0.147.16:5060) Callstate Change RINGING -> EARLY 2010-12-01 11:56:01.534727 [DEBUG] sofia.c:4576 Channel sofia/external/203170166 5@194.0.147.16:5060skipping state [early][183] 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:676 Send signal sofia/e xternal/2031701665@194.0.147.16:5060[BREAK] 2010-12-01 11:56:01.534727 [INFO] switch_cpp.cpp:1181 All Answered 2010-12-01 11:56:01.534727 [DEBUG] switch_core_session.c:1827 Application playba ck Requires media! pre_answering channel sofia/external/2031701665(a)194.0.147.16: 5060 EXECUTE sofia/external/2031701665@194.0.147.16:5060playback(/usr/local/freeswit ch/sounds/svc_sound_files/default_autoattendant.wav) 2010-12-01 11:56:01.537644 [DEBUG] switch_cpp.cpp:972 sofia/external/2031701665@ 194.0.147.16:5060destroy/unlink session from object Dialplan: sofia/external/2031701665@194.0.147.16:5060 Action set(effective_calle r_id_name=${404_tag}) Dialplan: sofia/external/2031701665@194.0.147.16:5060 Action bridge(${404_dial}) 2010-12-01 11:56:01.537644 [DEBUG] switch_core_state_machine.c:119 (sofia/extern al/2031701665@194.0.147.16:5060) State Change CS_ROUTING -> CS_EXECUTE 2010-12-01 11:56:01.537644 [DEBUG] switch_core_session.c:1057 Send signal sofia/ external/2031701665@194.0.147.16:5060[BREAK]

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Sangoma D100 usage
by Javier Gallart 02 Dec '10

02 Dec '10

Hello we got a Sangoma D100 transcoding card, and we like to get it working with our freeswitch. Freeswitch and the Sangoma card are running on the same server. The module is properly loaded: 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1102 Reading sangoma codec configuration 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1126 Found Sangoma RTP IP x.x.x.x 2010-12-02 07:46:33.393953 [NOTICE] mod_sangoma_codec.c:1193 Detected 1 and activated 1 Sangoma codec vocallo modules 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1207 Loading codecs, register='all', noregister='' 2010-12-02 07:46:33.393953 [DEBUG] mod_sangoma_codec.c:1239 Registering implementations for codec PCMU ... 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering implementations for codec PCMA ... 2010-12-02 07:46:33.394951 [DEBUG] mod_sangoma_codec.c:1239 Registering implementations for codec G729 ... 2010-12-02 07:46:33.402964 [CONSOLE] switch_loadable_module.c:946 Successfully Loaded [mod_sangoma_codec] 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding Codec PCMU 0 Sangoma PCMU 8000hz 10ms 64000bps ... 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding Codec PCMA 8 Sangoma PCMA 8000hz 20ms 64000bps ... 2010-12-02 07:46:33.402964 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 Sangoma G729 8000hz 40ms 8000bps ... 2010-12-02 07:46:33.403967 [NOTICE] switch_loadable_module.c:274 Adding API Function 'sangoma_codec' The codecs look also good: freeswitch@internal> show codecs codec,Sangoma G723,mod_sangoma_codec codec,Sangoma G729,mod_sangoma_codec codec,Sangoma GSM,mod_sangoma_codec codec,Sangoma PCMA,mod_sangoma_codec codec,Sangoma PCMU,mod_sangoma_codec What we don't know is how to actually use the card. We have forced a call with g729 in the A-leg and g711 in the b-leg but freeswitch doesn't know how to bridge them. I've tried unsuccessfully this config in the vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=Sangoma G729,Sangoma PCMU,Sangoma PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=Sangoma PCMU,Sangoma PCMA"/> Does anybody has any experience usgin this card? Thanks Javier

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Re: [Freeswitch-users] mod_callcenter and peridic announcements
by Raimund Sacherer 02 Dec '10

02 Dec '10

Hello Michael, thank you for pointing this out, it seems mod_fifo looks exactly like what we would need, well, apart from the asterisk call-strategies, but I think we can live with that. Best, Ray ----- Mensaje original ----- De: "Michael Collins" <msc(a)freeswitch.org> Para: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Enviados: Jueves, 2 de Diciembre 2010 2:08:54 GMT +01:00 Amsterdam / Berlín / Berna / Roma / Estocolmo / Viena Asunto: Re: [Freeswitch-users] mod_callcenter and peridic announcements mod_fifo supports periodic announcements. mod_fifo and mod_callcenter are two different ways to accomplish the same basic task, namely to queue up incoming calls and pass them to agents as efficiently as possible. -MC On Wed, Dec 1, 2010 at 7:44 AM, Raimund Sacherer < raimund.sacherer(a)logitravel.com > wrote: Hello Freeswitch List. We are evaluating switching our asterisk based call center (around 150 agents) over to freeswitch. Among other things I noticed that the mod_callcenter module does not have the possibility to use periodic announcements. Could these be implemented somehow ... like, with ESL or something else? I want to try out the Spice Technologies (or is it now OpenACD) callcenter solution, but this will be a further project as we first want to switch to freeswitch as soon as possible as we have instability problems which we want to get rid off. Thank you for your support, Ray -- Raimund Sacherer Dpto. de Sistemas Agencia de Viajes Online _______________________________________________ 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 _______________________________________________ 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 -- Raimund Sacherer España Dpto. de Sistemas Agencia de Viajes Online www.logitravel.com Edificio Logitravel Parcela 3B, Parc Bit Ctra. Palma - Valldemossa km 7,4 07121 Palma de Mallorca Tel 902 366 847 Fax 971 213 495

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dead call records
by Seven Du 02 Dec '10

02 Dec '10

Hi, I found a wired problem where left dead call records. originate {origination_uuid=1,hangup_after_bridge=false}user/1000 &park originate {origination_uuid=2,hangup_after_bridge=false}user/1001 &park uuid_bridge 1 2 uuid_transfer 1 -both park inline hupall show channels # show nothing show calls # the record still there, why the uuid is not 1 and 2 but a long uuid str? but there's no problem if not use origination_uuid. Also, there are no problems if no uuid_transfer. originate {origination_uuid=1,hangup_after_bridge=false}user/1000 &park originate {origination_uuid=2,hangup_after_bridge=false}user/1001 &park uuid_bridge 1 2 hupall I'm on last git, anyone can help take a look? http://pastebin.freeswitch.org/14681 Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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send fax
by Madovsky 02 Dec '10

02 Dec '10

Hi All, I'm currently playing with send a fax like this : /usr/local/freeswitch/bin/fs_cli -x "originate {originate_caller_id_number=9999999999999,fax_enable_t38=true,fax_enable_t38_request=true,absolute_codec_string=PCMU,nibble_account=9999999999999,nibble_rate=(lcr(1111111111111)*2.3)}sofia/gateway/trunk/1111111111111 &txfax(/usr/local/freeswitch/storage/FAX/test/test.tiff)" but it seems that the originate command line has chars length limit, so the command line is trunkated and give this result : 2010-11-28 12:15:24.929729 [ERR] mod_spandsp_fax.c:1019 Cannot send non-existant fax file [/usr/local/fre] also do I need to set tot true T38 the 2 vars if my trunkn is T38 compatible ? I noticed that send a fax has not a high percentage of success. Any help would be appreciated Franck

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mod_spandsp fax question
by Madovsky 02 Dec '10

02 Dec '10

Hi, is there any way to send multiple fax pages with one call with originate and &txfax() ? Thanks Franck

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loopback question
by Madovsky 02 Dec '10

02 Dec '10

Hi, what is the logical to send a fax outside from gateway with originate that sends the call to the default dialplan ? I tried ths : originalte {filetofax=myfax.tiff}loopback/0000numberToSendFax/default 99999999 and put this in default dialplan <extension name="send_fax"> <condition field="${sip_to_uri}" expression="^\+([2-9]\d{10,15})@$${domain}$"> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=false"/> <action application="ring_ready"/> <action application="set" data="absolute_codec_string=PCMU"/> <action inline="true" application="set" data="fax_file=${fax_file}"/> <action application="transfer" data="sendfax XML features"/> </condition> </extension> but doesn't work. Any ? Thanks Franck

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ringback on aleg during bleg_app
by Alexandre Fiori 02 Dec '10

02 Dec '10

Hi guys, When bridge_pre_execute_bleg_app is executing, aleg remain in silence. More specifically, let's say aleg is listening the ringback, bleg answer the call and start executing the bleg_app. After the call is answer by bleg, aleg immediately stops ringing and remain in silence while bleg_app is executing. How can I make aleg keep listening the ringback until bleg_app is finished and media (audio) is bridged? That is similar to the group_confirm_key=exec behavior, but I have to use bridge_pre_execute_bleg_app because it's not about confirming the call, it's about playing some audio files before media is bridged. The problem is, aleg should keep the ringback before bleg_app is finished. Thanks! -- Ship ahoy! Hast seen the While Whale? - Melville's Captain Ahab

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mod_callcenter and peridic announcements
by Raimund Sacherer 02 Dec '10

02 Dec '10

Hello Freeswitch List. We are evaluating switching our asterisk based call center (around 150 agents) over to freeswitch. Among other things I noticed that the mod_callcenter module does not have the possibility to use periodic announcements. Could these be implemented somehow ... like, with ESL or something else? I want to try out the Spice Technologies (or is it now OpenACD) callcenter solution, but this will be a further project as we first want to switch to freeswitch as soon as possible as we have instability problems which we want to get rid off. Thank you for your support, Ray -- Raimund Sacherer Dpto. de Sistemas Agencia de Viajes Online

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hangup_after_hook after txfax
by Madovsky 02 Dec '10

02 Dec '10

I'm trying to get the mod_span vaialbes after fax sent fax_success and fax_result_text with following in dialplan <action application="set" data="api_hangup_hook=system /usr/local/bin/php /usr/local/freeswitch/scripts/phptest.php ${dialed_extension} ${fax_file} ${fax_success} ${fax_result_text}"/> but these vars are empty. is there any example to get these vars after hangup ? Thanks Franck

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