August 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

how do I fix perl interface to streamFile?
by covici＠ccs.covici.com 15 Aug '11

15 Aug '11

I want to stream a file and have the dtmf digits move forward and backward in the file by a certain number of seconds. I first tried to set n inputcallback function and use uuid_fileman, but that does not work because when the digit is received, the file is actually stopped, so the seek is meaningless. Now the streamFile itself has provision for this, except that the perl interface just has the file and the sample count, omitting the callback function and its args. What do I have to change to putthe args in, so I can get this to work? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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odbc basic_calls
by Avi Marcus 15 Aug '11

15 Aug '11

I just upgraded FS since.. 7 weeks ago I think. Now: FreeSWITCH Version 1.0.head (git-9d98d49 2011-08-10 08-38-55 -0500) While testing the new build clean, when I hit F4 for show calls, I now get: freeswitch@internal> 2011-08-11 14:24:04.574797 [ERR] switch_core_sqldb.c:825 ERR: [select * from basic_calls where hostname='sip2' order by call_created_epoch] [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51 Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls' doesn't exist Aren't the core odbc tables supposed to be auto-created? -Avi

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please help!!! playing tone only when 3rd party joins in bridging conference
by ran zhang 15 Aug '11

15 Aug '11

I have a bridging conference established with 2 people in it, and i want to play a tone when 3rd party joins in. I can't set the conference's 'enter-sound' to play the tone since it will play the tone when first 2 people establish the bridging conference.

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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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