December 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

uuid_bridge on bridged channels
by Seven Du 01 Dec '10

01 Dec '10

Hi, Is it make sence to brige a bridged channel to another channel? I'm on FreeSWITCH Version 1.0.head (git-28f5d28 2010-11-09 14-11-02 -0500) And trying this: originate {origination_uuid=1,hangup_after_bridge=false,exec_after_bridge_app=park}user/1000 &park originate {origination_uuid=2,exec_after_bridge_app=park}user/1001 &park originate {origination_uuid=3,exec_after_bridge_app=park}portaudio/auto_answer &park and uuid_bridge 1 2 # Then it bridged, and then I did uuid_bridge 1 3 # 1 bridged to 3 and 2 parked, then uuid_bridge 2 3 # no bridges all channels parked Then I tried a few uuid_bridge between two of 1,2,3 until all channels hangup with destination out of order http://pastebin.freeswitch.org/14661 I was trying to move calls between live channels, should I unbridge all channels completely and bridge later? If so, is it possible to use the bridge app to bridge to a existing UUID instead of a dial string? (makes sense, since then we can use transfer bridge:other_uuid inline instead of uuid_bridge -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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Simple LUA not playing message issue
by Marc de Corny 01 Dec '10

01 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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Cant compile last git on FreeBSD
by korn 01 Dec '10

01 Dec '10

Hello, I,m trying to compile last git on FreeBSD 7.1 and receive this error in mod_spandsp: making all mod_spandsp Creating mod_spandsp_la-mod_spandsp_fax.lo Compiling mod_spandsp_fax.c ... cc1: warnings being treated as errors mod_spandsp_fax.c: In function 'configure_t38': mod_spandsp_fax.c:721: warning: implicit declaration of function 't38_set_fastest_image_data_rate' mod_spandsp_fax.c: In function 't38_gateway_on_consume_media': mod_spandsp_fax.c:1637: warning: implicit declaration of function 't38_gateway_rx_fillin' gmake[4]: *** [mod_spandsp_la-mod_spandsp_fax.lo] Error 1 gmake[3]: *** [mod_spandsp-all] Error 1 gmake[2]: *** [all-recursive] Error 1 configure and gmake in ./libs/spandsp are successful can anybody tell me, why I'm getting error WBR Evgeny

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mod_fifo or mod_callcenter
by abubacker 01 Dec '10

01 Dec '10

Dear all, I want to use mod_fifo or mod_callcenter to perform queuing operation , but I dont want this application to connect the agent and the customer I have the external script to do that but it should give me the dial string of the agent where the customer is likely to connect with. Is this possible or can we do this using some work around ? Thanks in advance ! -- Best regards, N.Abubacker , Associate system engineer , bk systems pvt ltd , Ph : 9144-43902701 Disclaimer: http://www.bksystems.co.in/email-policy

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FsGui Windows build
by Dmitry Bely 01 Dec '10

01 Dec '10

Hi, Has anyone succeeded in building FsGui for Windows from the current repository sources? The older one available at http://files.freeswitch.org/fsgui/fsgui.exe suffers form at least one serious problem (100% CPU load) that makes it hardly usable. Git logs hint that it may be fixed already. - Dmitry Bely

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DynamicRTP-Type-98
by Troy Anderson 01 Dec '10

01 Dec '10

I noticed a problem with calls originating from the mod_callcenter. The order of the codecs in the Media Description portion of the SDP is different for calls from mod_callcenter than when dialing the extension from another extension. Specifically, ITU-T G.711 PCMU is coming first on calls that work, and DynamicRTP-Type-98 is coming first on those that don't. A Polycom 450 is returning a 488 Not Acceptable Here message on the callcenter calls, and an Aastra 6731i has extremely choppy audio. If I tell the Aastra to only allow G722, all works as expected. I have my codecs set up in FS as global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM Can anyone explain what DynamicRTP-Type-98 is and why it would be in the SDP when I'm not specifying it in the codec list? Here's what the Media Description portion looks like when calling extension to extension: Media Description, name and address (m): audio 10140 RTP/AVP 0 98 99 9 8 3 101 13 Media Type: audio Media Port: 10140 Media Protocol: RTP/AVP Media Format: ITU-T G.711 PCMU Media Format: DynamicRTP-Type-98 Media Format: DynamicRTP-Type-99 Media Format: ITU-T G.722 Media Format: ITU-T G.711 PCMA Media Format: GSM 06.10 Media Format: DynamicRTP-Type-101 Media Format: Comfort noise Here's what the Media Description portion looks like when mod_callcenter initiates the call: Media Description, name and address (m): audio 12606 RTP/AVP 98 99 9 0 8 3 101 13 Media Type: audio Media Port: 12606 Media Protocol: RTP/AVP Media Format: DynamicRTP-Type-98 Media Format: DynamicRTP-Type-99 Media Format: ITU-T G.722 Media Format: ITU-T G.711 PCMU Media Format: ITU-T G.711 PCMA Media Format: GSM 06.10 Media Format: DynamicRTP-Type-101 Media Format: Comfort noise Thanks! -Troy

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