August 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS for SIP registrar server
by Dome Charoenyost 07 Aug '11

07 Aug '11

Dear All, I use FS for my wholesale solution. it's work fine over 2000 concurrent in peak hr. so in experience when i use FS for SIP registrar server with out ODBC (but use tmp for db path) i can handle 12,000 active in 1 server (25% CPU). All user send register packet every 3 minutes. Now i need system for handle 500,000 subscriber I have 2 choice 1. use Kamailio for SIP regitrar and use FS for routing and billing 2. still use multiple FS with ODBC So anyone can help? BG Dome C.

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Re: [Freeswitch-users] Radius/Authentication/Authorization
by Abid Saleem 06 Aug '11

06 Aug '11

Dear Tihomir, Any update and help on the below please. Abid From: abid_freeswitch(a)live.com To: tculjaga(a)gmail.com CC: neal(a)wanlink.com Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization Date: Wed, 31 Mar 2010 19:35:32 +0500 Dear Tihomir, Thank you very much for the configuration example but in which files to place these configurations. Please bear with me because I am new to FreeSwitch and if you could provide complete steps. Also when I compile FS and load the module mod_rad_auth in conf/autoload_configs/modules.conf.xml, I get an error while starting FS as follows. 2010-03-31 19:32:29.399466 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_rad_auth.so**/usr/local/freeswitch/mod/mod_rad_auth.so: undefined symbol: rc_conf_str** Please advise. Thanks. Regards-----------Abid Saleem --Forwarded Message Attachment-- From: tculjaga(a)gmail.com CC: neal(a)wanlink.com To: freeswitch-users(a)lists.freeswitch.org Date: Tue, 30 Mar 2010 21:42:03 +0200 Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization hello, here is an example <configuration name="rad_auth.conf" description="radius authentification module"> <settings>   </settings> <client> <param name="authserver" value="10.1.1.10:1812:gateway"/> <param name="dictionary" value="/usr/local/etc/radiusclient/dictionary.all"/> <param name="seqfile" value="/var/run/radius.seq"/> <param name="mapfile" value="/usr/local/etc/radiusclient/port-id-map"/> <param name="default_realm" value=""/> <param name="radius_timeout" value="3"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="bindaddr" value="*"/> </client> <vsas>   <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1" direction="in"/> <param name="h323-prompt-id" id="104" value="SERVICENUM" pec="9" expr="1" direction="in"/> <param name="Cisco-AVPair" id="1" value="TRANSACTIONID" pec="9" expr="1" direction="in"/> <param name="Calling-Station-Id" id="31" value="CALLINGNUMBER" pec="0" expr="1" direction="in"/> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0" direction="in"/> <param name="NAS-Port-Id" id="87" value="ISDN 3/0:D:14" pec="0" expr="0" direction="in"/> <param name="Login-User" id="1" value="1" pec="0" expr="0" direction="in"/>  <param name="BILING_MODEL" id="109" value="billing_model" pec="9" expr="0" direction="out"/> <param name="CREDIT_AMOUNT" id="101" value="credit_amount" pec="9" expr="0" direction="out"/> <param name="CURRENCY" id="110" value="currency" pec="9" expr="0" direction="out"/> <param name="PREFFERED_LANG" id="107" value="preffered_lang" pec="9" expr="0" direction="out"/> <param name="CREDIT_TIME" id="102" value="credit_time" pec="9" expr="0" direction="out"/> <param name="H323-IVR-IN:DIRATION" id="1" value="h323_ivr_duration" pec="9" expr="0" direction="out"/> <param name="RADIUS_RETURN_CODE" id="103" value="return_code" pec="9" expr="0" direction="out"/>  </vsas> </configuration> in the dialplan you need to trigger auth as: <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> there are two behaviours: 1. authorize the call according to username&pass and dialed number - if authorized, the radius server returns credit time towards the dialed number 2. authorize the call according to username&pass - if authorized, the radius server returns the current account balance <extension name="unitest_rad-ANI-auth"> <condition field="destination_number" expression="^601$"> <action application="log" data="INFO PRIJE RAD_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=${return_code}"/>  </condition> </extension> <extension name="unitest_rad-ANI-balance"> <condition field="destination_number" expression="^602$"> <action application="log" data="INFO PRIJE RAD_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 inline="true" application="set" data="DIALED_NUMBER=16094191500"/> <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> will update the wiki by the end of the week. you have enough information for now. Tihomir. On Tue, Mar 30, 2010 at 1:52 PM, Abid Saleem <abid_freeswitch(a)live.com> wrote: Hi Neal and other Contributors to FS, I recieved an answer on the list that mod_rad_auth is ready. I upgraded FS to download and install it by "make current" and it is successfully built and installed. Could you please outline the detailed steps to do all this configuration. If it is still not ready, is there any other method that somebody has already implemented like perl scripts etc. Any help in this regard is badly required. THanks for your cooperation. Regards----------Abid SaleemSr. Product Manager Hotmail: Powerful Free email with security by Microsoft. Get it now. _________________________________________________________________ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969

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outgoing SIP profiles setup
by lloyd thomas 06 Aug '11

06 Aug '11

Hi All, I am trying to create a new sip profile as my provider does not use sip registrations, but uses IP address and port number to identify wether a connection should be allowed. I am currently using the 'internal' profile to make calls '<action application="bridge" data="sofia/internal/$1(a)80.40.150.150"/>', but this is causing a couple of strange anomalies. I have created a new xml file in sip_profiles and use 'sofia profile phisys-2circles start' to add it. But I get the following errors and the profile does not show in 'sofia status' errors----------------------------------------------- 2011-08-06 17:38:28.620292 [NOTICE] sofia_reg.c:2775 Added gateway ' example.com' to profile 'phisys-2circles' 2011-08-06 17:38:28.620292 [WARNING] sofia.c:1898 Ignoring duplicate gateway 'example.com' 2011-08-06 17:38:28.624397 [WARNING] sofia.c:1898 Ignoring duplicate gateway 'example.com' 2011-08-06 17:38:28.625645 [NOTICE] sofia.c:3999 Started Profile phisys-2circles [sofia_reg_phisys-2circles] 2011-08-06 17:38:28.643631 [DEBUG] sofia.c:1471 Creating agent for phisys-2circles 2011-08-06 17:38:28.800994 [ERR] sofia.c:1533 Error Creating SIP UA for profile: phisys-2circles 2011-08-06 17:38:28.800994 [NOTICE] sofia_glue.c:5198 deleted gateway example.com I have added phisys_2circles.xml to the sip_profiles directory and the contents are here http://pastebin.freeswitch.org/16998 Any advice welcome as I am at a lost what to do next. Thanks Lloyd T

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SIP auth challenge error
by cogs66 06 Aug '11

06 Aug '11

Hello All First post here and a newb...... I have just updated the latest GIT and now seeing the following in the CLI. Does anyone know how to fix this or is it nothing to worry about? [WARNING] sofia_reg.c:1339 SIP auth challenge (REGISTER) on sofia profile 'internal' for xxx from ip -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-auth-challenge-error-tp66… Sent from the freeswitch-users mailing list archive at Nabble.com.

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outboundsocket mode, bridge and bind_digit_action doesn't work
by Michael Ricordeau 05 Aug '11

05 Aug '11

Hi, I think I'm doing something wrong but I don't find a way to fix my problem : when executing bind_digit_action in an outbound socket and doing a bridge, digit action binding is not executed (no log in Freeswitch) when doing same thing in XML dialplan it works (I found log in Freeswitch). I have checked logs and in both cases, digit realm is set : Digit parser DPTOOLS: binding 00/test/0 callback: 0xb6ba1b00 data: 0x82eafe8 For outbound socket, here what I'm doing with netcat : nc -l -v 8084 and the commands I passed to outbound socket : connect divert_events on sendmsg call-command: execute execute-app-name: answer sendmsg call-command: execute execute-app-name: bind_digit_action execute-app-arg: test,00,exec:log,NOTICE TEST sendmsg call-command: execute execute-app-name: bridge execute-app-arg: user/1000 With XML Dialplan below, same binding/bridge works !!!!! : <extension name="test"> <condition field="destination_number" expression="^0000$"> <action application="answer"/> <action application="bind_digit_action" data="test,00,exec:log,NOTICE TEST"/> <action application="bridge" data="user/1000"/> </condition> </extension> Expected result is when A leg presses "00", a log notice "TEST" is printed in the logger but only works with XML dialplan. In outbound socket mode, I can hear the digits pressed on B leg but binding is not executed (no log notice "TEST") Thanks Michaël

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Hangup hooks on B legs
by Isaac Jurado 05 Aug '11

05 Aug '11

Hi, I've been playing with the "api_hangup_hook" dialplan setting. However, it only seems to be executing when the A-leg (originator) hangs up. Is there any way to execute such a hook from all the potential B-legs of the call? Thanks in advance. -- Isaac Jurado Internet Busines Solutions eConcept

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Cisco Presence
by Peder 05 Aug '11

05 Aug '11

Has anybody ever managed to get presence working on the Cisco 79x1 series phones? I know the 79x0 phones don't support it, but the x1 and newer should. It works on the SPA, but that isn't what I am looking for.

1 0

Does FreeSWITCH supports Googles VP8 (WebM) Codec
by Markus Siebeneicher 05 Aug '11

05 Aug '11

Hello folks from FreeSWITCH, i am using FreeSWITCH with the conference system BigBlueButton, made by people from canada. It works pretty fine, so far. Due to the need that we would like to support more than 500 users in a single video/audio session, we are in interest to use the best codecs. So i listen to VP8 from Google. Does FreeSWITCH support it anyway? I couldn't find something in the FAQ's. I listen something about ffmeg, is it in this abstraction module? Thanks for a short reply and thanks for the good work! Kind regards, Markus

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My new web utility for FreeSWITCH
by Valery Kalinin 05 Aug '11

05 Aug '11

FreeSWITCH channel viewer Web-based PHP utility for online view FreeSWITCH channels. All calls are displayed _without_ refreshing the page. Program uses comet-style transport, without using ajax & flash. Tested on: Internet Explorer 8.0, FireFox 4.0.1 Some columns are made thin. Click on him for explode. Any column collapse/explode on click. Download & installation guide: https://sites.google.com/site/freeswitched/home/channel-viewer How this work (data = list of calls): client (browser) server (PHP script) --------------- ------------------------------------------------- send data ---> get list of calls (call FreeSWITCH) compare data apply changes <--- send only changes of data cicle while php script timeout (max_execution_time): - get list of calls (call FreeSWITCH) - each 1 sec - compare data apply changes <--- - send only changes of data restart script <--- after php script timeout send command to restart send data ---> ...repeat... All these tricks are made for traffic minimization. Tests and feature requests are welcome. Thanks.

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Disable Allow Events in Fs
by Simon Leck 05 Aug '11

05 Aug '11

Hi Everybody, I am trying to disable allow events in FS. Can some give me some guideline on how to do it. Thanks in advanced. Many thanks Simon

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