July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Discovered problem with PocketSphinx build?
by Bryan Smart 14 Aug '11

14 Aug '11

Tried a build with a latest GIT, with PocketSphinx enabled, and get: Creating mod_pocketsphinx.la... /bin/sed: can't read /usr/src/freeswitch/libs/pocketsphinx-0.7/../sphinxbase/src /libsphinxbase/libsphinxbase.la: No such file or directory quiet_libtool: link: `/usr/src/freeswitch/libs/pocketsphinx-0.7/../sphinxbase/sr c/libsphinxbase/libsphinxbase.la' is not a valid libtool archive cat: .libs/mod_pocketsphinx.log: No such make[5]: *** [mod_pocketsphinx.la] Error 1 I don't pretend to fully understand the build system, but the makefile seems to be trying to go to libs/sphinxbase for files, when that directory doesn't exist. The currently downloaded version of sphinxbase has the version appended to the directory name, as in sphinxbase-0.7. The Sphinx readme warns that the root should be named sphinxbase. I did: ln -s sphinxbase-0.7 sphinxbase and the build finished without problems. PocketSphinx works fine on a hardware phone using ULAW, but, when I use a softphone with 16Khz Speex, it doesn't. Instead, I get this 2011-07-27 21:43:14.651803 [WARNING] mod_pocketsphinx.c:147 Can't open speech model /usr/local/freeswitch/grammar/model/wsj1. The only item in the model directory is communicator. I thought wsj1 was the old model. What in Freeswitch would try to use wsj1 over communicator based on the codec/rate? The mod_pocketsphinx wiki page says that 16Khz is a supported sampling rate, so I'm guessing that there is a special reason why Speex doesn't work, or else that this is a bug. Bryan

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Fidelio
by Nandy Dagondon 09 Aug '11

09 Aug '11

hi everybody, anyone working on interfacing FS with Fidelio Hotel PMS? i can't find the FIAS protocol/specs online. is this freely available? tks, nandy

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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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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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Help setting up SIP reg
by lloyd thomas 03 Aug '11

03 Aug '11

*Hi I need a little help setting up a SIP registration for a provider that does not use auth.* *All I have is info below.* ** * * SBC/Proxy IP: 80.40.150.150:5060 Authentication: Trusted IP – 88.221.85.33 Assigned DDI: 01869******, 01869****** DTMF Method: RFC2833 Status: Live No. of trunks: 2x Session Timer: 1800 Profile*:* Generic (35060) Apparently the following is used for * [vibe] type = friend host = 80.40.150.150

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conference caller id number&name
by ran zhang 03 Aug '11

03 Aug '11

I want when the bridging conference is inviting someone to start the conference, the caller is shown as the real caller id number&name rather than whats stored in 'caller-id-number' and 'caller-id-name' in conference.conf.xml.

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mod_rtmp question
by Madovsky 02 Aug '11

02 Aug '11

I use freeswitch in ODBC mode. in sofia endpoint if I type "sofia status profile internal" I get all registrations from all nodes but in rtmp endpoint "rtmp status profile internal reg" I get registrations of the local node. is it a normal behaviour ? if yes which rtmp command can give the same result of sofia with ODBC ? Thanks Franck

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ESL: not bein able to determine when SIP gateway is down when originating a call
by Anton VG 02 Aug '11

02 Aug '11

I'm trying to catch an error, in case I would dial wrong (non existent) gateway (intentionally!) I'm running ESL outbound listener, subscribing to all events, if I do bgapi 'originate' to a live gateway - there are normal events flow, and I can track what is happening. But if I issue originate to a gateway, which is not configured or simply down - there are no any events fired. I only have an error on FS console 2011-05-14 20:58:13.072927 [ERR] mod_sofia.c:4044 Invalid Gateway following by 2011-05-14 20:58:13.072927 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] But HOW to catch the given in ESL? sofia.c seems just does not have event code for that cases if (profile_name && !profile_found) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "No Such Profile '%s'\n", profile_name); >status = SWITCH_STATUS_FALSE; } logically there should be a proper way to determine that gateway is bad in my ESL dialplan, by catching the proper event/reply/whatever, For the moment i did trick: esl.api('sofia status gateway GatewayWhichIsDown') When in production, and there is more than a single route, there will be plenty of cases, when you dial a bad gateway, so there should be a way for ESL dialplan to determine that a gateway is not callable for a moment, the reason WHY and to retry with another one. The trick above is bad, since: 1. blocking api query, before evey single gateway call attempt. 2. Gateway maybe known in UP state, but the state is stale, in dial in fact will go to DOWN gateway. So, ESL dialplan will screw in that case Any clue?

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Still having problems making calls to phones behind double nat
by Rob Hutton 01 Aug '11

01 Aug '11

I am still unable to figure this one out and would appreciate any help. I have two phones at a remote location that can call out through the switch and have two way audio, but no one can call into them. The switch returns a 606 User Not registered even though show registrations seems to indicate that they are. I don't see anything in the sip trace to indicate that it actually tried to contact the destination user, but INFO keepalives are going back and forth... Configuration and sofia log are here: http://pastebin.freeswitch.org/16784

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unable to connect to freeswitch server using x-lite
by Dhairya Vora 01 Aug '11

01 Aug '11

I have installed freeswitch on centos ----------------------------------------------------- [root@localhost ~]# uname -r 2.6.18-238.19.1.el5 ------------------------------------------------------ I have installed x-lite on a different machine in same lan. when i try to connect x-lite to freeswitch with valid id, password, request times out. any idea to solve this? -- Dhairya Vora(newbie)

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