April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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api_hangup_hook and hupall
by Dome Charoenyost 04 Aug '11

04 Aug '11

I use api_hangup_hook for do something when call hangup. but when i use hupall command from CLI api_hangup_hook do nothing. Is it bug ? Dome C.

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Complete wholesale app in freeswitch
by David Villasmil 21 Jul '11

21 Jul '11

Hello Guys, I'm finishing a "complete" wholesale application created on freeswitch and I was wondering whether it would be a good idea to put it up on the wiki. I just don't know how. Features include all the following parameters configurable via web interface: - Profile creation based on server IP where traffic is received. You can have multiple IPs, system will create multiple profiles/diaplans so it can differentiate. - i.e. offer to the same customer a "gold" routing on IP1 and cheap routing on IP2 - Customer add/modify/delete - IP source - Rates for client routes based on areacode - Prepaid or postpaid. - When cutomer balance is 0, no more calls are allowed. - limit max channels - Media by-pass - When by-passed, customer and provider will exchange RTPs directly. Else, server will be in the middle. - Provider add/modify/delete - costs for provider routes based on areacode - limit max channels - Routing based on areacode, gives great granularity. - Routes can be assigned multiple gateways/providers which can in turn be distributed based on weigth. Includes overflow to next configured GW. - Basic financial report generation (totals) by customer/provider - Basic traffic ASR/ACD report (totals) by cutomer/provider - Basic user administration. (No access level, only total access) - CDR export to csv file. I also have a prepaid card app... no web interface on that one though... Thanks all David

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mod_pocketsphinx
by roy 19 Jun '11

19 Jun '11

Hi, Trying to get mod_pocketsphinx working from the current git tree, FreeSWITCH version: 1.0.head (git-2629a57 2010-06-16 22-42-15 +0200) however, I got this error during startup: <snip> 2010-06-17 11:18:08.032645 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_pocketsphinx.so **/usr/local/freeswitch/mod/mod_pocketsphinx.so: undefined symbol: ngram_model_get_counts** </snip> Lib paths were already defined in /etc/ld.so.conf as: include /usr/local/freeswitch/lib /usr/local/freeswitch/mod # ls -al mod/mod_pocketsphinx.* -rwxr-xr-x 1 root staff 1467 Jun 17 10:49 mod/mod_pocketsphinx.la -rwxr-xr-x 1 root staff 905769 Jun 17 10:49 mod/mod_pocketsphinx.so Was wondering if someone was able to get this working. Manually downloading the current pocketsphinx and sphinxbase and renaming them into 0.4.99 and rebuilding didn't help either. Would appreciate if anyone can point me into the right direction. TIA. Roy

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Music On Hold
by Chia-Yen Wu 16 Jun '11

16 Jun '11

Hello, I am using freeswitch as an IVR server. I would like to be able to put the call on hold and play music while performing some other tasks in the lua. when the task is complete , the music stop and the IVR continue but i cant find a way to do that, is there any function can do "music on hold" in dialplan ? Thank you

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mod_fsv, record and playback options
by William Kendi ... 27 May '11

27 May '11

Greetings, Currently, I am trying to record and play videos through FreeSWITCH using the "mod_fsv" module. Some options like timeouts and interrupts by dtmf digits seems to be lacking in the "mod_fsv" module and I cant even stop the video recording using the "break" application. The video recording stop when the user disconnects. Anyone has any ideas about how to stop the FSV video recording without disconnecting the user? Thanks in advance.

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Tuning Up Freeswitch
by Antonio Teixeira 24 May '11

24 May '11

Hello List. I'm currently integrating an IVR in python together with freeswitch using mod_python and ESL and my life has been well until ... The flow of calls went over 80 simultaneous calls. Now freeswitch starts sending packets with huge delays ( even when establishing the call , mainly the 200 ) and firing up the IVR with tons of delay up to 20 seconds. So i searched the wiki forums and mailing list: Put freeswitch on a diet , trimmed modules.conf Played with the ulimit stuff. Played with the IVRS to reduce load to a minimum and i was able to squeeze more 5 calls of performance. The problem is : Top shows top - 16:14:33 up 35 days, 8:15, 3 users, load average: 1.92, 1.76, 1.78 Tasks: 133 total, 1 running, 132 sleeping, 0 stopped, 0 zombie Cpu(s): 1.4%us, 3.3%sy, 0.0%ni, 94.6%id, 0.0%wa, 0.3%hi, 0.5%si, 0.0%st Mem: 8193336k total, 1639156k used, 6554180k free, 177208k buffers Swap: 19534904k total, 0k used, 19534904k free, 1062272k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 31361 yadayada 20 0 716m 164m 9628 S 73 2.1 155:17.85 freeswitch Freeswitch goes up to 150 % and puff there goes the MOS down to 0. Some basic System Info : Debian 6.0 ( i heard the timming module is affected by Debian , but if the CPU % gets lower than 95% everything will be more stable) Python 2.5 2 x Intel(R) Xeon(R) CPU E5506 @ 2.13GHz 8 GB of Ram as you can see 94 % of the "Cpu Power" is sleeping :\ It appears freeswitch is only capable of using let's say "one cpu"/thread ?? Do you guys recommend simply starting more instances or redoing the IVR stuff. Hope you guys can help me out. Thanks António Teixeira

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T.38 via UPDATE request
by Peter P GMX 19 May '11

19 May '11

Hello, we have the following problem. 2 Fax machines are communicating via Freeswitch. One is externally attached via a Telco who is able to handle T.38. The other one is attached locally. When 2 Fax machines start syncing each other, the Telco sends a SIP UPDATE message with T.38 SDP, as it detects fax during the fax negociations. Freeswitch answers with an SIP OK message back to the telco, and I can see the T.38 SDP on the debug console of freeswitch. Then nothing happens any more until one of fax machines detects timeout. We have set proxy-media to true. However is was done during call setup when both machines communicated with G711 SDP. The UPDATE message was commited by FS to the telco, but was not sent to the other fax, so I think in this case Freeswitch is supposed to transcode between T.38 and G711 which it cannot do, as we know. How can I overcome this scenario? Is this a defect, should freeswitch send the UPDATE message to the other fax? Or is there a workaround? Best regards Peter Here's the UPDATE message: UPDATE sip:mod_sofia@82.115.xx.xxx:5080 SIP/2.0. Call-ID: 5283fe4e-334f-122d-d1b9-001517956764. Contact: <sip:212.91.xxx.xxx:5060>. Content-Type: application/sdp. CSeq: 16340063 UPDATE. From: <sip:06912345678@sip.telco.de>;tag=00-08135-017e041a-4d21f6037. Max-Forwards: 31. Route: <sip:82.115.96.165;lr;ftag=pH663F4S02erm>. To: "030987654321" <sip:030987654321@82.115.xx.xxx>;tag=pH663F4S02erm. User-Agent: Cirpack/v4.41e (gw_sip). Via: SIP/2.0/UDP 212.91.xxx.xxx:5060;branch=z9hG4bK-178C-1923B31. Content-Length: 300. . v=0. o=cp10 125551618103 125551618105 IN IP4 212.91.xxx.xx. s=SIP Call. c=IN IP4 212.91.xxx.xx. t=0 0. m=image 6860 udptl t38. a=sendrecv. a=T38FaxVersion:0. a=T38MaxBitRate:9600. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:1000. a=T38FaxMaxDatagram:200. a=T38FaxUdpEC:t38UDPRedundancy.

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freeSWITCH segfault on openwrt
by Woody Dickson 18 May '11

18 May '11

Hi, I manage to get freeSWITCH to compile and run on openwrt, but then freeSWITCH keeps crashing when the number of calls reaches 25 consistently. Content of the core file returns nothing when opened with gdb. Does anyone have any clue as to why this is happening and how to fix it? thanks, woody

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problem with transferring calls when using ivr
by bogusmaster＠o2.pl 14 May '11

14 May '11

Hi all ! I'm a freeswitch newbie taking care of installation that I've recently taken over from a colleague who left our company. Currently I'm having a problem with transferring calls that went thru ivr menu. The call scenario looks like this: external number A calls our external number E. Then, in ivr menu it selects some option. After that A is connected with internal number I. When internal number I tries to transfer call from A to another internal number I2, call hangs up. If I turn the ivr off, as it is commented in the excerpt from /conf/dialplan/public/01_freephone_inbound.xml below, transferring calls works fine. <extension name="general_inbound"> <condition field="destination_number" expression="^(freephone_XXXX_in)$"> <action application="info"/> <action application="transfer" data="100 XML default"/>  </condition> </extension> Below is my ivr script: //==================================================================================== // Function :: Debug output //==================================================================================== function Debug(text) { var uuid_last5 = session.uuid.substr(-5) console_log("info", "["+uuid_last5+"] "+text+"\n"); } //==================================================================================== // Function :: Callback handlers for collectInput // Returning false from these handlers causes the digit collection // to "unblock". Depending on what behavior you want to achieve // you might want to unblock on any input or on a specific digit. //==================================================================================== function Get_DTMF_Stop_On_Any(session, type, data, arg) { Debug("DTMF: "+data.digit+""); arg.digits += data.digit; return false; } //===================================================================================================================== //===================================================================================================================== //===================================================================================================================== //===================================================================================================================== //==================================================================================== // Main Section //==================================================================================== var dtmf_digits = new Object(); session.answer(); Debug("Playing welcome message..."); session.streamFile("/opt/freeswitch/sounds/mlife/ivr/welcome.wav"); dtmf_digits.digits = ""; var repeat = 0; while (session.ready() && dtmf_digits.digits == "" && repeat < 3) { //------------------------------------------------------ // Flush digits, ask for menu selection //------------------------------------------------------ dtmf_digits.digits = ""; session.flushDigits(); Debug("Playing announcement..."); session.streamFile("/opt/freeswitch/sounds/mlife/ivr/ann.wav", Get_DTMF_Stop_On_Any, dtmf_digits); if(dtmf_digits.digits == "") { session.collectInput(Get_DTMF_Stop_On_Any, dtmf_digits, 3000); } Debug("Menu Choice Selected = [" + dtmf_digits.digits + "]"); if (dtmf_digits.digits == "") { repeat++; if (repeat > 2) { Debug("Max tries reached"); break; } } else { if (dtmf_digits.digits == "1") { session.execute("execute_extension", "100 XML default"); } else if (dtmf_digits.digits == "2") { session.execute("execute_extension", "200 XML default"); } else if (dtmf_digits.digits == "3") { session.execute("execute_extension", "401 XML default"); } else if (dtmf_digits.digits == "4") { //session.execute("transfer", "ivr-support"); Debug("Playing warning message about recording..."); session.streamFile("/opt/freeswitch/sounds/mlife/ivr/support.wav"); session.execute("execute_extension", "tech-support XML default"); } } } // User exceeded the ivr prompt limit, say bye-bye // session.streamFile("conference/mlife/8000/conf-goodbye.wav"); session.hangup(); Any hints would be greatly appreciated. Regards, Bart -- Using Opera's revolutionary email client: http://www.opera.com/mail/

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