August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

e164.org
by Ivan C Myrvold 27 Aug '08

27 Aug '08

I thought I would register my telephone number with e164.org, so that anyone can call my number with SIP. I am not going to have to make a new profile for this, as the calls will come in on port 5060, right? I see the test call coming in on my FreeSWITCH (which is on a public ip and not behind NAT), but FreeSWITCH returns "Authentication Required", which kills the test call. Do anyone have a setup for this they would share? Ivan

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Problem with xml_cdr_mod
by Juan Backson 27 Aug '08

27 Aug '08

Hi, I am trying to get Freeswitch to post CDR to HTTP using xml_cdr_mod. Here is the error in console: 2008-08-28 01:13:11 [DEBUG] mod_sofia.c:264 sofia_on_hangup() Channel sofia/internal/1012(a)192.168.1.400 hanging up, cause: NORMAL_CLEARING 2008-08-28 01:13:11 [ERR] mod_xml_cdr.c:189 my_on_hangup() Got error [405] posting to web server [http://192.168.1.400:10000/CDR] 2008-08-28 01:13:11 [ERR] mod_xml_cdr.c:200 my_on_hangup() Unable to post to web server, writing to file What is causing this error? What does that error means and how to resolve it? Thanks, JB

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firing custom event in Lua
by Juan Backson 27 Aug '08

27 Aug '08

Hi, I tried to fire custom event with the following example: e = freeswitch.Event("message"); e:add_body("mymsg","abcd") session:sendEvent(e) But I am getting this error from freeswitch console: 2008-08-25 05:49:33 [ERR] mod_lua.cpp:176 lua_parse_and_execute() ...r/local/freeswitch/scripts/script.lua:61: attempt to call method 'add_body' (a nil value) stack traceback: ...r/local/freeswitch/scripts/script.lua:61: in main chunk Does anyone know what is wrong with my script? Thanks, JB

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Nested limits in xml-curl
by Tim Panton 27 Aug '08

27 Aug '08

I'm using xml-curl to dynamically generate dialplan, (very neat feature by the way) but I'm running into a problem where the limits kick in too soon Here is what I'm doing (edited to remove secrets etc) -> <action application="limit" data="$$ {domain} 192.67.4.33 2"/> <action application="limit" data="$$ {domain} 00441612370661 1"/> <action application="set" data="call_timeout=300"/> <action application="bridge" data="sofia/blahxxxx"/> My question is - Am I allowed to set 2 different limits on the same bridge action (and a timeout) from within xml-curl ? Tim.

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How to mix 3 way voice ?
by Lee JJ 27 Aug '08

27 Aug '08

Hello : Here is the testing dialplan. <extension name="070_gogotalk"> <condition field="destination_number" expression="^(070[0-9].*)">  <action application="bind_meta_app" data="3 a a javascript::calltest.js" /> <action application="bridge" data="sofia/gateway/sip.ttn.net/$1"/> </condition> </extension> While caller call out a 070xxx user , during talk caller press "*3" , FW execute calltest.js ### calltest.js ### // Create new_session n_sess = new Session() ; res = n_sess.originate(n_sess, "sofia/inter2/1527%210.243.126.72" ); // wait for it ... answer n_sess.waitForAnswer(10000); // IF everybody is ready, then bridge our current session & the new_session //session.ready() && new.session ready if ( n_sess.ready() ) { //bridge(session, n_sess); //session.execute("threeway", sv_uuid ) ; //n_sess.execute("threeway", n_sess.uuid ) ; session.execute("conference", "3001@default" ) ; n_sess.execute("conference", "3001@default" ) ; } No matter I tried "bridge" , "threeway" or conference , there always one leg hangup. 2008-08-27 12:22:37 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/inter2/1527 [5 fe74280-953d-4aba-a738-f0b8e01bfc60] 2008-08-27 12:22:38 [NOTICE] sofia.c:2167 sofia_handle_sip_i_state() Ring-Ready sofia/inter2/1527! 2008-08-27 12:22:41 [NOTICE] sofia.c:2491 sofia_handle_sip_i_state() Channel [sofia/inter2/1527] has been answered 2008-08-27 12:23:06 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/external/0702058899 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] How to put all 3 legs into a conference room ? Should I have to specify each uuid ? Thanks !

2 1

latest freeswitch for windows
by xbipin 27 Aug '08

27 Aug '08

can any1 plz compile the latest freeswitch for windows with the ability to generate certificates for TLS protocol as i am just unable to do it with my little knowledge for C and visual studio. basically i looking for he latest windows freeswitch with TLS, as in plugins etc included for generating SSL certifiates. if any1 can do it then i can host it and provide very good bandwidth for others to download. -- View this message in context: http://www.nabble.com/latest-freeswitch-for-windows-tp18751293p18751293.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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multiple sip domains
by Tamas Cseke 27 Aug '08

27 Aug '08

Hello, I have a problem with multiple domains. [WARNING] sofia_reg.c:1251 sofia_reg_parse_auth() can't find user [cstomi(a)192.168.1.37] You must define a domain called '192.168.1.37' in your directory and add a user with the id="cstomi" attribute and you must configure your device to use the proper domain in it's authentication credentials. [DEBUG] sofia_reg.c:705 sofia_reg_handle_register() send challange for [10(a)192.168.1.37] I have multiple domains in the directory, and put the user under the proper domain. <domain name="test"> <user id="cstomi"> <params> <param name="password" value="cstomi"/> </params> </user> </domain> But I can't configure the client device (another FS) to use this domain in INVITEs, REGISTER is fine. I saw the to domain is used in the realm of the challange, what determine the domain in the directory, isn't it? my gateway settings: <gateway name="test"> <param name="realm" value="test"/> <param name="username" value="cstomi"/> <param name="password" value="cstomi"/> <param name="extension" value="inbound"/> <param name="proxy" value="192.168.1.37:5060"/> <param name="expire-seconds" value="600"/> </gateway> REGISTER: It seems the to host is made from the realm...I'm not sure.. So it works, I don't get the WARNING above. but in INVITE the to host is seems to be the "proxy". I guess if I could change the to host, authorization would work...How could I do this? Thanks any help, Tamas

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Re: [Freeswitch-users] Problem with xml_cdr_mod [SOLVED]
by Juan Backson 27 Aug '08

27 Aug '08

Hi, This is due to missing doPost() in my servlet. It is fixed now. Please ignore. Thanks, JB On Wed, Aug 27, 2008 at 5:03 PM, Juan Backson <juanbackson(a)gmail.com> wrote: > Hi, > > I am trying to get Freeswitch to post CDR to HTTP using xml_cdr_mod. > > Here is the error in console: > > 2008-08-28 01:13:11 [DEBUG] mod_sofia.c:264 sofia_on_hangup() Channel > sofia/internal/1012(a)192.168.1.400 hanging up, cause: NORMAL_CLEARING > 2008-08-28 01:13:11 [ERR] mod_xml_cdr.c:189 my_on_hangup() Got error [405] > posting to web server [http://192.168.1.400:10000/CDR] > 2008-08-28 01:13:11 [ERR] mod_xml_cdr.c:200 my_on_hangup() Unable to post > to web server, writing to file > > What is causing this error? What does that error means and how to resolve > it? > > Thanks, > JB >

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anybody done nuance mrcp?
by Shlomi Hazan 27 Aug '08

27 Aug '08

Does anybody know how to get some softphone (xlite) to talk to nuance and get the text out? Best Regards, Shlomi.

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Loadbalance and failover
by ram 27 Aug '08

27 Aug '08

Hi iam just thinking new setup replacing Asterisk with Freeswitch in my lab setup I want to users to register with Openser(opensips) and PSTN/ VoIP Termination with FS users------Openser------Number of FS Boxes Does any one deployed sucessfully the above said setup If so kindly recomend me how to do this in best manner ram

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