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Fax Issue
by Peder 01 Dec '10

01 Dec '10

Anybody know how to debug a failed fax issue? LOG: http://pastebin.freeswitch.org/14687 VERSION: FreeSWITCH Version 1.0.head (git-13e8de4 2010-12-01 13-07-47 -0600) FAX.CONF.XML: <configuration name="fax.conf" description="FAX application configuration"> <settings> <param name="use-ecm" value="true"/> <param name="verbose" value="true"/> <param name="disable-v17" value="false"/> <param name="ident" value="SpanDSP Fax Ident"/> <param name="header" value="SpanDSP Fax Header"/> <param name="spool-dir" value="/tmp"/> <param name="file-prefix" value="faxrx"/> </settings> </configuration> DIALPLAN: <include> <extension name="fax_receive"> <condition field="destination_number" expression="^(6083)$"> <action application="set" data="api_hangup_hook=system ${base_dir}/scrip ts/emailfax.php $1 '${caller_id_name}-${caller_id_number}' test(a)test.com /tm p/${uuid}.rxfax.tiff"/> <action application="answer"/> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/> <action application="log" data="CRIT Success - [${fax_success}]"/> <action application="log" data="CRIT Rate - [${fax_transfer_rate}]"/> <action application="hangup"/> </condition> </extension> </include> I've got 3-4 different people trying to fax to me and every one fails with a similar log to what I have above. I had it working at one point in time, but I get about 2 faxes a month, so I have no clue when it broke or what broke it. The log just seems to show that negotiation failed, but I can't tell why. Thanks. Peder

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FreeSWITCH "Staff" Conference Call on Thursday Dec 2
by Michael Collins 01 Dec '10

01 Dec '10

Hello all, We are planning a 30 minute staff conf call for tomorrow, December 2 at 1PM EST / 10AM PST. If you are willing and able to assist with the FreeSWITCH infrastructure maintenance and are ready to roll up your sleeves and do some work then please join us tomorrow. We will use the FreeSWITCH public conference: SIP: 888(a)conference.freeswitch.org PSTN: 1-919-386-9900 Skype: skypiax5 Some of the things we will be discussing: Who can do what? (i.e. what are you good at and what would you like to learn how to do) FS infrastructure overview: what we have, what we are planning to do, etc. and stuff that needs management: * git * JIRA * Fisheye * Drupal sites (freeswitch.org, cluecon.com, ostag.org) * Mailing lists (-users, -dev, etc.) * other servers (db, conf box, etc.) We have a lot to go over, but initially we just need to get everything out on the table. If you want to help please email me off list and provide me your contact information and what your skills are. We are filling out this list: http://wiki.freeswitch.org/wiki/Community_Staff Thank you all for supporting FreeSWITCH! -Michael

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freeswitch transfer call problem
by Михаил Салтанов 01 Dec '10

01 Dec '10

HelloI have some problem with freeswitch call transfer by python script. ----------------------------------------------- import os from freeswitch import * import psycopg2 import socket def handler(session, args): print 'ready to transfer' number_to_transfer = session.getVariable('number_') print number_to_transfer session.setAutoHangup('false') session.execute('set', 'hangup_after_bridge=false') session.execute('set', 'continue_on_fail=true') session.transfer( number_to_transfer, "XML", "default") Description of my problem Phone # 1 call to Phone #2 Phone #2 input some digits for transfer incoming call to phone #3 [phone #1-->phone#3] Phone #1 hangup, Phone 2 connect to phone # 3 where the my mistake? When I make transfer by xml extensions all works fine С уважением Михаил Салтанов

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SNMP - FS_cli -x syntax
by Francis Trevor 01 Dec '10

01 Dec '10

I have created the activechannels script as recommended in another post to export active calls so that they may be displayed in a NMS. This is basically a "fs_cli -x distinct_channels" However, is it possible to show channels for a particular tech prefix? Or rather put, what is the syntax for the fs_cli -x show channels like <999>. Whereas 999 would be the tech prefix. Am I on the right track for building the logic to show active calls per customer? TGF

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Record File
by Will Smith 01 Dec '10

01 Dec '10

hi,I am trying to record a call after bridging the 2 legs. 2 parties can talk/listen to each other. There is an audio file created, but just silence, and always stopped at 44bytes. I just wonder if there is any other setup that I missed.I used 2 ways to record the call, and they both gave the same result:1- session.execute("record_session", "foo.wav"); 2- rtn = session.recordFile("foo.wav", true, "", 500,500,3); Thank you for your help. Will

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/usr/local/freeswitch/mod/mod_gsmopen.so: cannot open shared object file: No such file or directory
by Anshu Sah 01 Dec '10

01 Dec '10

Hi, I was following steps of http://wiki.freeswitch.org/wiki/GSMopen to Install GSMOpen, OS : UBUNTU Server v10 FreeSWITCH Version 1.0.6 gcc (Ubuntu 4.4.3-4ubuntu5) 4.4.3 After Completing Last step ----------------------------------------------------------------------- cd /usr/src/freeswitch.trunk/src/mod/endpoints/mod_gsmopen/configs/ cp gsmopen.conf.xml /usr/local/freeswitch/conf/autoload_configs/ vi /usr/local/freeswitch/conf/autoload_configs/gsmopen.conf.xml ----------------------------------------------------------------------- When I run ----------------------------------------------------------------------- /usr/local/freeswitch/bin/freeswitch *load mod_gsmopen* ----------------------------------------------------------------------- *ERROR* ----------------------------------------------------------------------- 2010-11-29 10:32:43.064571 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_gsmopen.so **/usr/local/freeswitch/mod/mod_gsmopen.so: cannot open shared object file: No such file or directory** ----------------------------------------------------------------------- I tried to search mod_gsmopen.so, But didn't found anywhere :( Please guide. Regards Anshu Sah

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how to use mod_curl in Lua on window platform ???
by dhiraj bharti 01 Dec '10

01 Dec '10

Hi, Anybody have any idea how to user cURL in Lua on window platform ? Thanks in Advance -- Dhiraj bharti DBS Technologies Pvt. Ltd 9990448967

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FreeSWITCH Conference Call Today
by Michael Collins 01 Dec '10

01 Dec '10

Hello all, Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_01 We'll be doing another community documentation session but first we'll have Chad Phillips, aka hunmonk, talking to us about his Jester Mail project. Talk to you in a bit. -MC

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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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