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Mod event socket error
by Leonardo P. Bidinoto 15 Aug '11

15 Aug '11

Hi Guys, Im getting a lot of errors in my test machines, like these: 2011-08-15 10:12:39.433325 [CRIT] mod_event_socket.c:378 Lost 244 events! 2011-08-15 10:12:47.689947 [CRIT] mod_event_socket.c:378 Lost 32 events! 2011-08-15 10:12:48.770116 [CRIT] mod_event_socket.c:378 Lost 3 events! 2011-08-15 10:12:50.771600 [CRIT] mod_event_socket.c:378 Lost 54 events! 2011-08-15 10:12:50.791728 [CRIT] mod_event_socket.c:378 Lost 9 events! 2011-08-15 10:12:51.012380 [CRIT] mod_event_socket.c:378 Lost 39 events! 2011-08-15 10:12:51.012380 [CRIT] mod_event_socket.c:378 Lost 1 events! What this kind of error means? -- Leonardo Pires Bidinoto Voice Technology www.voicetechnology.com.br

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Conference calls through web-interface with moderation using FreeSWITCH?
by Alec Taylor 15 Aug '11

15 Aug '11

Good Morning, I have been researching this for a while, basically I'd like to have a website with the following functionality: • One-click call-in to show (after setting username, best-case scenario: sign-in through Drupal, use that name for conference-call) • Web-interface only (Flash/Flex, Javascript/JQuery or Java), without any additional software/addons/plugins to install • Moderation: host of conference call can quieten/mute or even kick people from the conference call if they're being rowdy So far I have setup an IceCAST server, broadcasting through edcast in an mp3 stream. Viewers of my website can now listen-in on the /radio/ sub-page. How do I setup the aforementioned [3] features using FreeSWITCH? — Do I need [Free, Open-Source] products other than FreeSWITCH to get this done, if so, which? Thanks for all suggestions, Alec Taylor

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Running Post-Bridge scripts
by Alec Glassford 15 Aug '11

15 Aug '11

Hi, I need to run a LUA script after a Bridge to write call information to a MySQL DB. The problem is the script is only running intermittently, maybe 1 in 10 calls. I have tried the exec_after_bridge_app command, I've tried uuid_bridge_continue_on_cancel, but it neither works. I can provide output if necessary. I am bridging FreeTDM Sangoma A101 E1 ISDN30 calls to a SIP Provider, I do a DB lookup to find the correct Presentation number then bridge the call, below is from my Dialplan: <condition field="destination_number" expression="^(\d{11})$"> <action application="lua" data="lookup02.lua"/> <action application="set" data="effective_caller_id_number=${Presentation}"/> <action application="set" data="effective_caller_id_name=${Presentation}"/> <action application="log" data="INFO Dialling $1 presenting ${Presentation} from dblookup"/> <action application="set" data="uuid_bridge_continue_on_cancel=true"/> <action application="set" data="exec_after_bridge_app=lua"/> <action application="set" data="exec_after_bridge_arg=endcall01.lua"/> <action application="bridge" data="sofia/gateway/ProviderZZZ/$1" /> <action application="lua" data="endcall01.lua"/> <action application="hangup" /> </condition> Any help much appreciated Alec Glassford efuse Tel: 0844 847 9707 Mob: 07540 417395 Fax: 0844 847 9708 www.efuse.co.uk<http://www.efuse.co.uk/> This is an email from efuse, IT Solutions providers: www.efuse.co.uk<http://www.efuse.co.uk/> P Save a tree...please don't print this e-mail unless you really need to Its contents are confidential and legally privileged and it is intended only for the use of the addressees named above. If you are not an addressee you must not read it and must not use any information contained in it nor copy it nor inform any person other than Efuse Solutions or the addressees of its existence or contents. If you have received this email and are not a named addressee please delete it and notify support(a)efuse.co.uk<mailto:helpdesk@efuse.co.uk> Please note that Efuse Solutions nor the sender accepts any responsibility for viruses and that it is your responsibility to scan any attachments. No contractual obligations may be established on behalf of Efuse Solutions by means of email communication.

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Playback .wma/.wmv
by Duvid Rottenberg 15 Aug '11

15 Aug '11

I would like to playback into a channel a windows media stream (mms streaming .wma or the audio track of a .wmv), however it appears that the playback command does not support this. I have loaded the mod_shout module but it appear this also doesn't support this format. Is there any module available that supports this format? Thanks, Duvid Rottenberg

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Open Source Billing for FreeSWITCH to get public soon ..
by Muhammad Naseer Bhatti 15 Aug '11

15 Aug '11

Hello, I thought this would be the right time to let everyone know we are going to release open source billing for FreeSWITCH. Completely developed in PHP/MySQL. Some of the major features includes: Both Pre-Paid and Post-Paid model Multiple administration access levels Multiple reseller level Easy rate/price management Route management Separate user interface to view their CDR(s) and billing information Authentication by IP/ANI and SIP registration Codec management for both user and switch CDR statistics Gateway statistics Admin/Reseller/User management Switch management from 1 GUI Balance and payment information ... and much more. We need volunteers for testing. Please drop an email --[ vbilling [at] digitallinx.com ]-- if anyone is interested. Thanks.

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Secure connection
by Chrisbware 15 Aug '11

15 Aug '11

Just to have your opinion on it: I've a PHP site on a server, connected to Freeswitch (mod_event_socket) on another server. Which method would you recommend to protect data trasmitted between two servers (they travel on internet)?

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mod_rtmp & flex client
by Matthew Fong 14 Aug '11

14 Aug '11

I am trying to get the mod_rtmp to work with the flex client, but I am receiving the following errors when I try to make a test call 2011-08-14 06:04:32.744112 [NOTICE] rtmp_sig.c:121 Sent connect reply 2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream (1) 2011-08-14 06:04:38.764112 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3) 2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream (2) 2011-08-14 06:04:38.764112 [NOTICE] switch_channel.c:904 New Channel rtmp/default/5000 [18578182-a702-49bc-9590-c53ec2d16b72] 2011-08-14 06:04:38.764112 [ERR] rtmp_sig.c:305 Couldn't create call. 2011-08-14 06:04:38.764112 [INFO] mod_dialplan_xml.c:336 Processing <0000000000>->5000 in context public 2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:194 rtmp/default/5000 has executed the last dialplan instruction, hanging up. 2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:196 Hangup rtmp/default/5000 [CS_EXECUTE] [NORMAL_CLEARING] 2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1346 Session 1 (rtmp/default/5000) Ended 2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1348 Close Channel rtmp/default/5000 [CS_DESTROY] 2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:159 Sending audio 2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3) 2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:274 Got publish on stream 2. 2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3) I am using a version checked-out from yesterday, Ubuntu 10.4 64-bit and flash debug version. Does anyone know what I am doing wrong? Or has anyone gotten it to work? Also I tried the FS conference call test using the hosted version on conference.freeswitch.org and it seems the first few seconds of audio are distorted. Is there a way to fix this? Thanks... mod_rtmp seems very promising. --matt

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Record-Route Header Rejection
by Loy Glenn 14 Aug '11

14 Aug '11

Accepted Header Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=gK0653a65f> Rejected Header Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=SDpr0p702-3521913535-971555%uukeghgcjiudgieijkcuu0u> So it must be the ftag...What is wrong with it? Or am I completely off the mark? FreeSWITCH returns SIP/2.0 400 Bad Record-Route Header FreeSWITCH Version 1.0.head (git-6d1d4a9 2011-08-09 16-48-58 -0500) -- You Rock! Your E-Mail Should Too! Signup Now at Rock.com and get 2GB of Storage! http://connections.rock.com/user/displayUserRegisterPage.kickAction?as=1167…

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incompatible destination because of crypto on incoming calls
by Anestis Mavro 14 Aug '11

14 Aug '11

Hi, Suddenly (latest git) something happened with the incoming calls. All calls have crypto enabled and they get dropped with "INCOMPATIBLE DESTINATION" I have TLS disabled in vars.xml but still I see in debug: Sofia_glue:2900 Set Local Key [1 AES_CM_128_HMAC_SHA1_32 inline: ...] Show channels and show calls show now a lot of calls, but they don't exist; they don't get cleared. Anybody any idea where this can come from? Thank you

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Mod_rad_auth issue for FS working with FreeRadius server
by fieldpeak 14 Aug '11

14 Aug '11

Hello Gurus, i met a issue when using mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works with freeradius server+mysql for AAA, the details is below, Could anyone give any hints, Thanks in advance. i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however, when i dialed 601 to trigger the dial plan, the console show errors, it looks "h323-conf-id" is not in the directory, then i tried to add this attribute to the dictionary, however, it does not help, in the wiki, it mentioned the rad_auth.conf.xml contains <param name="dictionary" value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did not find the file "dictionary.all" at that directory, so i use dictionary. BTW, the freeradius server + mysql works well. console errors: EXECUTE sofia/internal/1001(a)124.193.106.104 auth_function(in , in 38516060333, in 003282, out AUTH_RESULT) 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate initial structure. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed configuration. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver := 127.0.0.1:1812:gateway. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary := /usr/local/etc/radiusclient/dictionary. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile := /var/run/radius.seq. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile := /usr/local/etc/radiusclient/port-id-map. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set default_realm := . 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_timeout := 3. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_retries := 2. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_deadtime := 0. 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr := *. 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius: User-Name: 38516060333 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius: User-Password: 003282 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius: Called-station-Id is empty, ignoring... 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle attribute: h323-conf-id 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not found in dictionary 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending radius packet. 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured during RADIUS Authentication(RC=-1) 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured during radius authorization. EXECUTE sofia/internal/1001(a)124.193.106.104 log(INFO AUTH_RESULT=) 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202 AUTH_RESULT= EXECUTE sofia/internal/1001(a)124.193.106.104 log(INFO billing_model=) 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202 billing_model= EXECUTE sofia/internal/1001(a)124.193.106.104 log(INFO credit_amount=) 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202 credit_amount= EXECUTE sofia/internal/1001(a)124.193.106.104 log(INFO currency=) 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202 currency= EXECUTE sofia/internal/1001(a)124.193.106.104 log(INFO preffered_lang=) 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202 preffered_lang= added below in the dictionary(/usr/local/etc/radiusclient/dictionary): ATTRIBUTE h323-conf-id 1008 string dial plan: <extension name="unitest_rad-ANI-auth"> <condition field="destination_number" expression="^601$"> <action application="log" data="INFO Before Auth "/> <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/> <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/> <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>  <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/> <action inline="true" application="set" data="USERNAME=38516060333"/>  <action inline="true" application="set" data="PASSWD=003282"/>  <action application="sleep" data="2000"/> <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> <action application="log" data="INFO AUTH_RESULT=${AUTH_RESULT}"/> <action application="log" data="INFO billing_model=${billing_model}"/> <action application="log" data="INFO credit_amount=${credit_amount}"/> <action application="log" data="INFO currency=${currency}"/> <action application="log" data="INFO preffered_lang=${preffered_lang}"/> <action application="log" data="INFO credit_time=${credit_time}"/> <action application="log" data="INFO h323_ivr_duration=${h323_ivr_duration}"/> <action application="log" data="INFO return_code=${return_code}"/>  </condition> </extension> radius_cdr.conf.xml: <configuration name="radius_cdr.conf" description="RADIUS CDR Configuration"> <settings>  <param name="dictionary" value="/usr/local/freeswitch/conf/radius/dictionary"/>  <param name="radius_retries" value="3"/>  <param name="radius_timeout" value="5"/>    <param name="acct_server" value="127.0.0.1:1813:testing123"/> </settings> </configuration> the FS version: FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400) Regards, Charles

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