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Query Mod_skypopen
by Zuhair Raza 11 Jun '10

11 Jun '10

Hi Skype module cannot be loaded on Centos 5.5, Did anyone experienced it???? -- Regards, Zuhair Raza

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Receiving SEGV on Outbound ESL
by lakshmanan ganapathy 11 Jun '10

11 Jun '10

Hi all, I'm trying to do the following scenario. I've created an Inbound socket. I issue the following command in the Inbound ESL api originate {origination_caller_id_name='lakshmanan',origination_caller_id_number=227,call=1,user=2,type=test,id=1001,refid=847,enabled=0,ignore_early_media=true,originate_timeout=30}user/1006 202 XML default My dialplan looks like follows: <context name="default"> <extension name="outbound_soc"> <condition field="destination_number" expression="^.*$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="bypass_media=false"/> <action application="set" data="ignore_early_media=true"/> <action application="set" data="exec_after_bridge_app=park"/> <action application="socket" data=" 0.0.0.0:8447 async full"/> </condition> </extension> </context> So once the user 1006 answers the call, it will connect to an Outbound socket. And here is my program in perl for controlling the Outbound ESL. use lib '/root/freeswitch/libs/esl/perl/'; require ESL; use IO::Socket::INET; my $ip = "localhost"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', Proto => 'tcp', Listen => 1, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; for(;;) { printf "Going to wait for clients to connect\n\n"; my $new_sock = $sock->accept(); my $pid = fork(); if ($pid) { close($new_sock); next; } &registerSignals(); my $host = $new_sock->sockhost(); my $fd = fileno($new_sock); printf "Before making a ESL connection $fd\n"; my $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s\n", $uuid, $info->getHeader("caller-caller-id-number"); print "BYE\n"; close($new_sock); } sub registerSignals() { foreach ( keys %SIG ) { $SIG{$_} = 'sig_Handler'; } } sub sig_Handler() { my $handle = shift; print "Got signal $handle\n"; if($handle eq "CHLD") { wait(); } } But as soon as the call comes to the Outbound ESL program, it receives SIGSEGV. The child program prints the following: Before making a ESL connection 4 Got signal SEGV Can any one please tell me why this is happening?? And one more thing. If I execute the following API, it works fine as expected. api originate {origination_caller_id_name='lakshmanan',origination_caller_id_number=227,call=1}user/1006 202 XML default The child prints as follows: Before making a ESL connection 4 Connected call 01200ab6-73bf-11df-af54-55cf82f02c9b, from 227

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Help with Sofia
by José Vera 10 Jun '10

10 Jun '10

Hi there, Again, I'm trying to get freeswitch working, It gave me credentials and the info needed to get the service working but when the user connects to it I'm getting this error: http://img821.imageshack.us/img821/4562/sofia.png , Anyone knows how to disable sofia AUTH or set it to allow anyone to use it easy? wich steps I need to do (will add this info to a guide I'm working on for non experienced users like me) :) Thanks in advance, Jorel

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JavaScript control Transfer
by David Swardstrom 10 Jun '10

10 Jun '10

I have been working on a small Javascript application to determine the specific Javascript application to execute based on the DNIS. The goal is: * Provide an entry in a dialplan public XML file to invoke the application. * Use a database lookup (ODBC) using DNIS. * Fail if the DNIS is not found (hangup the session) * If the DNIS is found: ** use another entry in the database to determine ** the working application and any static parameters. * Invoke/Transfer to/Activate the other application. It is fairly simple to do all but the first step. I have been having problems trying to get the last step to work. What I would like is something like this: session.execute("transfer", transfer_string); Where transfer_string is the string from the Database. Or where transfer_string is the string from the DB plus some fixed options. I would prefer not to have to add entries into the dialplan/pubic/00_xx.xml file but will if I have to. I did notice an option called "inline" which potentially avoid this. I suspect part of my problem is not quite understanding the transfer syntax. If I ever get this running properly, I will provide a Wiki example of a simplified version of this (ie, without some of work specific code). Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom

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Weekly FreeSwitch Conference call June 9 @ 1pm EST
by Norman Tomlins 10 Jun '10

10 Jun '10

Weekly FreeSwitch Conference call June 9 @ 1pm EST Hi Everyone, I would just like to remind everyone that we have our weekly conference call on June , 9th @ 1pm EST. During this conference call Darren Schreiber will talk about mod_nibblebill. Information on mod_nilbble can be found here: http://wiki.freeswitch.org/wiki/Mod_nibblebill Information on Weekly Conference call: http://wiki.freeswitch.org/wiki/FS_weekly_2010_06_09 Norman Tomlins ---------------------------------------------------------------------------- Voice Network Inc. http://www.VoiceNetwork.ca Unlimited Incoming US DID's - $ 3.95 - No Setup Fees Unlimited Incoming Canadian DID's - $ 5.95 - No Setup Fees ----------------------------------------------------------------------------

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Callback with bypass media
by Vitalii Colosov 10 Jun '10

10 Jun '10

Hi, Could you please advice if this callback scenario can be achieved? Freeswitch calls one number via external provider. After it answered, it gets the digits and dials another number via external provider. After second phone is answered, it bridge both sessions. Is it possible that from this moment media will flow directly from one sip phone to another, bypassing freeswitch? And if yes, which command (in Lua preferably)? I tried to use the following code after getting the digits from first phone: ... calling_session:setVariable("bypass_media", "true"); called_session = freeswitch.Session(some_called_string, calling_session); ... But as per log, the INVITE to the second phone contains FreeSWITCH's IP address in SDP section, and not first phone's IP address. Looks like bypass_media is not working as I would expect in this case. Thanks, Vitalie

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Re: [Freeswitch-users] uuid_media hangs
by Kristian Kielhofner 10 Jun '10

10 Jun '10

That's completely ridiculous. -- Kristian Kielhofner http://blog.krisk.org ------------------------------ *From*: freeswitch-users-bounces(a)lists.freeswitch.org < freeswitch-users-bounces(a)lists.freeswitch.org> *To*: freeswitch-users(a)lists.freeswitch.org < freeswitch-users(a)lists.freeswitch.org> *Sent*: Thu Jun 10 12:30:42 2010 *Subject*: Re: [Freeswitch-users] uuid_media hangs > are very keen to actually follow all these silly RFC I agree, usually RFC are created first by university so by students... ----- Original Message ----- *From:* Anthony Minessale <anthony.minessale(a)gmail.com> *To:* freeswitch-users(a)lists.freeswitch.org *Sent:* Thursday, June 10, 2010 12:09 PM *Subject:* Re: [Freeswitch-users] uuid_media hangs The funny part is you are willing to go to all this effort to try to make us correct it because you know the other people wont. This type of finger pointing to RFC is typical to providers so now I have to do free work to investigate it for you. And I bet you are paying them money and not us, oh well, typical.....Don't worry I won't hold it against you. The philosophical question here is weather or not to send a requires timer in the reinvite, so ok. I can try to look it up for you but from a logical point of view, once you establish in the original invite that you are going to use session timers, every reinvite constitutes a timer refresh, so once you have agreed in the original invite to use timers it would stand to reason to require the timer option in every new re-invite since it represents a timer refresh. But, let's face it... Who cares right? its a meaningless param in an REINVITE. Most sip implementation wing it on most of these packet parsers. The sofia developers are very keen to actually follow all these silly RFC you are trying to quote and they have been relaxing it nonstop ever since because nobody really follows any rules in SIP they just use them to avoid having to patch their code. Especially in a deployed appliance in a telco. They are really telling you they can't support it because the code in their sip gateway fails when it sees this and they are too lazy to change it so they pray that some RFC will let them off the hook. Trust me, we've had worse. One place refused our calls because they did not like the syntax of our user agent string...... So bottom line is I would trust the default behavior of sofia-sip over most things any day but if they are really doing something wrong the problem would be in the sofia-sip library not in FreeSWITCH so I am not sure where to tell you to look in the code to correct it because we use that code asis and avoid hacking into it a all costs. Try using grep into the sofia-sip lib for the strings you know are meaningful. P.S All the event related things are fixed now, there was a small speed bump along the way this week. On Thu, Jun 10, 2010 at 4:34 AM, peely <freeswitch(a)peely.com> wrote: > > Hi, > > It's not a hijacked thread, I created this with an issue around uuid_media, > initially it was hanging, then in the latest git it receives a 4xx from the > far end due to the Requires: timer support bot being supported in the > reinvite transaction. I did anecdotally mention problems with events, which > I subsequently pulled off to another thread. > > I was looking at RFC 4028, in the examples it shows the refresh event not > providing the Required: timer header even for the refresh of the original > invite transaction, just the Supported: timer. There's nothing in RFC 3725 > regarding sesison timers in 3PCC but as the reinvite for the media > adjustment is a new transaction, I don't think it needs the Requires: timer > option to persist the timer support on the original dialogue. This is > certainly the behaviour of the previous equipment I had on this > interconnect > and it operates using session timers and 3PCC in several other environments > too. > > I'm not brilliant with C, but if you could please point me to the right > approach to removing the Requires: timer field in the header I'd be happy > to > test it and let you know if it causes any problems with rienvites across > various carriers and other platforms. > > > > Cheers, > > > Neil. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p51… > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:+19193869900 ------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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LUA Callback - Help
by Aloysius Lloyd 10 Jun '10

10 Jun '10

Hi All, I am trying to build a callback application using LUA. I have the following setup. - Dial the DID number - Hangup - Callback lua initiate the call after 10 secods The above working great. But when I answer the call ...the wav file never play on the channel.What I am doing wrong? <extension name="CallBack"> <condition field="destination_number" expression="^6471231234$"> <action application="set" data="api_hangup_hook=luarun callback.lua ${caller_id_number}"/> <action application="hangup" /> </condition> </extension> *callback.lua* os.execute("sleep " .. tonumber(10)); pin_number_test="0511"; out_gateway1="sofia/gateway/voipms/1"; message_dir="/usr/local/freeswitch/sounds/en/us/callie/custom/callback/"; number_to_call= argv[1]; session = freeswitch.Session("{bypass_media_after_bridge=true,ignore_early_media=true,originate_timeout=40}"..out_gateway1..number_to_call); session:setAutoHangup(false); if ( session:ready() ) then session:sleep(3000); session:streamFile(message_dir.."enter-dest.wav"); end Thanks Lloyd

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Lua script hangup detect
by Chia-Yen Wu 10 Jun '10

10 Jun '10

Hello, i tried to use Lua script to replace xml macro in dialplan, but I found out that Lua wont terminate if client hangup, ,so the session is still on but client is already hangup, is there a way to avoid this ?

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LUA - session:waitForAnswer - Error
by Aloysius Lloyd 10 Jun '10

10 Jun '10

Hi All, In javascript the session.waitForAnswer(10000); working without any problem. But in Lua session:waitForAnswer(10000); throwing the following error. *2010-06-10 10:54:21.029607 [ERR] mod_lua.cpp:182 Error in waitForAnswer (arg 2), expected 'CoreSession *' got 'number' stack traceback:* What is the correct argument for this function in LUA. In wiki this function description is not available. Any help appreciate. Thanks Lloyd

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