October 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Mod Event Socket
by Will Smith 03 Nov '10

03 Nov '10

Hi,I am new to FS, and trying to get the mod event socket installed and running. I have FS running, with SIP account, can dial in/out via gateway.Now I want to dial in, FS will send some info to client browser, here is one question, does this work over the internet, or just local ? Info sent could be the uuid, so that client browser could decide to bridge the call, send to IVR or transfer ...Please give me some guide line how to set this up.I added to dialplan/default.xml----- <extension name="socketTest"> <condition field="destination_number" expression="^444$"> <action application="socket" data="127.0.0.1:9090 async full"/> </condition> </extension>----- got the php sample file: <?php /** * Based loosely on the NET_Server code in PEAR. * This is only an example - considerable additional work is needed * Specifically, the code in the handleConnection method should be * handled in a subclass * * */ class Message { var $properties = array(); var $content = null; } class EventSocketListener { var $host; var $port; var $sock; var $is_parent = true; var $clientInfo; var $clientFD; var $connectionContext = array(); function &create($port, $host = "localhost") { $esl = new EventSocketListener; $esl->port = $port; $esl->host = $host; if (!function_exists('socket_create')) { return PEAR::raiseError('Sockets extension not available.'); } return $esl; } function start() { if (($this->sock = socket_create(AF_INET, SOCK_STREAM, SOL_TCP)) === false) { echo "socket_create() failed: reason: " . socket_strerror(socket_last_error()) . "\n"; } if (!socket_set_option($this->sock, SOL_SOCKET, SO_REUSEADDR, 1)) { echo 'Unable to set option on socket: '. socket_strerror(socket_last_error()) . PHP_EOL; } if (socket_bind($this->sock, $this->host, $this->port) === false) { echo "socket_bind() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n"; } if (socket_listen($this->sock, 5) === false) { echo "socket_listen() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n"; } // Dear children, please do not become zombies pcntl_signal(SIGCHLD, SIG_IGN); // wait for incmoning connections while (true) { // new connection if(($fd = socket_accept($this->sock))) { $pid = pcntl_fork(); if($pid == -1) { return PEAR::raiseError('Could not fork child process.'); } // This is the child => handle the request elseif($pid == 0) { // this is not the parent $this->_isParent = false; // store the new file descriptor $this->clientFD = $fd; $peer_host = ""; $peer_port = ""; socket_getpeername($this->clientFD, $peer_host, $peer_port); $this->clientInfo = array( "host" => $peer_host, "port" => $peer_port, "connectOn" => time() ); $this->handleConnection(); socket_shutdown($this->clientFD, 2); socket_close($this->clientFD); } else /* Parent does nothing */ { } } } } function handleConnection() { $fd = $this->clientFD; //first, read headers & setup a state for this connection $line = ""; socket_write($fd, "CONNECT\n\n"); do { $line = socket_read($fd, 2048, PHP_NORMAL_READ); if (trim($line) == "") break; //we got a header, we need to add it to the list($key, $value) = explode(":", $line); $key = trim($key); $value = trim(urldecode($value)); $this->connectionContext[$key] = $value; } while ($line != ""); // print_r($this->connectionContext); $this->callConnected(); exit(); } function processMessages($returnOnReply = false) { $fd = $this->clientFD; $result = new Message(); $props = array(); while (true) { do { $line = @socket_read($fd, 2048, PHP_NORMAL_READ); if (socket_last_error($fd) == 104) return null; if ($line == null || $line == FALSE || trim($line) == "") break; //we got a header, we need to add it to the message list($key, $value) = explode(":", $line); $key = trim($key); $value = trim(urldecode($value)); $props[$key] = $value; } while ($line != ""); $result->properties = $props; if (isset($props['Content-Length'])) { $length = $props['Content-Length']; print("Reading content - $length\n"); $data = socket_read($fd, $length); $result->content = $data; } if (isset($props['Content-Type'])) { $type = $props['Content-Type']; if ($returnOnReply && ($type == "command/reply" || $type == "api/response")) { return $result; } else if ($type == "text/event-plain") //only plain events for now { $this->handleEvent($result); } } else { print("UNKNOWN MESSAGE: \n"); print_r($result); } } } function invokeCommand($command) { //Send the command print("Invoking: $command\n"); $this->sendCommand($command); // Wait for the response $result = $this->processMessages(true); return $result; } function sendCommand($command) { $fd = $this->clientFD; socket_write($fd, trim($command) . "\n\n"); } /*-----------------------------------------------------*/ /* Abstract Methods - should move to subclass*/ function callConnected() { print_r($this->connectionContext); print("----------------\n"); $result = $this->invokeCommand("log DEBUG"); print_r($result); $result = $this->invokeCommand("event plain ALL"); print_r($result); $this->processMessages(false); print("DONE PROCESSING MESSAGES"); print_r($this->connectionContext); } function handleCommandResponse($response) { print("Recieved Unhandled Response:\n"); print_r($response); } function handleEvent($event) { print("Recieved Unhandled Event:\n"); print_r($event); } } // create a server that forks new processes $server = &EventSocketListener::create(9090); // start the server $server->start(); ?>--------( this is the original file, not perfect sample) I tried to run this, and got error with auth.Also, I modify the even_socket_conf.xml in autoload_configs/ change listen-ip to 0.0.0.0 , port = 9090 , disable password What did I miss? Thankyou for your helpWill

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ivr terminitor key?
by Seven Du 01 Nov '10

01 Nov '10

Hi, For ivr menus that have digit-len > 1, is there a way to specify a terminator key, say #, to stop digits collecting immediately? Default behavior tends to wait until a timeout. Thanks. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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Probably obvious but not to me - sofia hangup
by general＠bestoronto.com 01 Nov '10

01 Nov '10

Hi, I am on freeswitch / centOS: Linux pbx 2.6.18-194.11.3.el5.centos.plus #1 SMP It is a fairly straight out of the box freeswitch config, I am not so good with XML yet. Issue is the line to my single SIP phone will drop after about 8 minutes, no matter if talking or not. What might be the cause of this? Seems like a timeout or lack of ACK somewhere? Thanks for the help.... Mark B

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conference profile question
by covici＠ccs.covici.com 31 Oct '10

31 Oct '10

Hi. I have some conferences setup, not using the default profile. Now if the conference calls someone using the dial command, he seems to come in with the default profile -- any variable or other thing I can set so he comes in with the profile I called the conference originally with by the conference app, or a better way to accomplish this? thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Loss of first second of media
by Fraser Redmond 30 Oct '10

30 Oct '10

I've been trying out a few new termination providers, and am finding that the first second of media is being lost. I get early media (ringing), then when the call is answered the first half a second to a second is lost, then the rest of the call is fine. The engineers at one of the providers ran a sip trace and a full packet trace, and it looked like everything was being delivered from the pstn to my server. They said that none of their other customers have reported this problem before. I've tested connecting to pstn's in several different countries, with the same results. I've used a couple of other termination providers in the past (1 still in current use) which have been fine, but now have 2 that are losing the first part of the media. So it seems like it's specific to Freeswitch, with some subset of conditions for some providers. This is running on Ubuntu on a Amazon AWS server, in case that helps. Any ideas how to narrow it down further? Cheers, Fraser

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mod_cepstral for Windows build -> Change language
by Christian Knoblauch 30 Oct '10

30 Oct '10

Hi, Is there a method to "dynamically" change the language for cepstral TTS at runtime ? Thanks & Rewgards Christian

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Problems making a receiveing calls with mod_jingling
by Rob Hutton 30 Oct '10

30 Oct '10

I am trying to get mod_dingaling working, and I am sure I am doing something stupid, but I can't figure out what it is... Relevant parts of call log: Dialplan: sofia/sipinterface_1/1001(a)192.168.1.200 Regex (PASS) [dingaling_1_pattern_1] destination_number(94045550100) =~ /^91{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$/ break=on-false Dialplan: sofia/sipinterface_1/1001(a)192.168.1.200 Action set(prepend=1) Dialplan: sofia/sipinterface_1/1001(a)192.168.1.200 Action set(hangup_after_bridge=true) Dialplan: sofia/sipinterface_1/1001(a)192.168.1.200 Action bridge(dingaling/GoogleTalk/+${prepend}4045550100(a)voice.google.com) 2010-10-13 12:15:41.427591 [DEBUG] switch_core_state_machine.c:119 (sofia/sipinterface_1/1001(a)192.168.1.200) State Change CS_ROUTING -> CS_EXECUTE ... EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 hash(insert/192.168.1.200-spymap/1001/87e31e76-56ff-47f2-b210-a316c3191a44) EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 hash(insert/192.168.1.200-last_dial/1001/94045550100) EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 hash(insert/192.168.1.200-last_dial/global/87e31e76-56ff-47f2-b210-a316c3191a44) EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 set(RFC2822_DATE=Wed, 13 Oct 2010 12:15:41 -0400) 2010-10-13 12:15:41.431854 [DEBUG] mod_dptools.c:1024 sofia/sipinterface_1/1001(a)192.168.1.200 SET [RFC2822_DATE]=[Wed, 13 Oct 2010 12:15:41 -0400] EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 set(prepend=1) 2010-10-13 12:15:41.431854 [DEBUG] mod_dptools.c:1024 sofia/sipinterface_1/1001(a)192.168.1.200 SET [prepend]=[1] EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 set(hangup_after_bridge=true) 2010-10-13 12:15:41.431854 [DEBUG] mod_dptools.c:1024 sofia/sipinterface_1/1001(a)192.168.1.200 SET [hangup_after_bridge]=[true] EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 bridge(dingaling/GoogleTalk/+14045550100(a)voice.google.com) 2010-10-13 12:15:41.432855 [DEBUG] mod_dingaling.c:1727 Unknown Profile! 2010-10-13 12:15:41.432855 [DEBUG] mod_dingaling.c:704 Terminate called from line 1728 state=CS_NEW 2010-10-13 12:15:41.432855 [NOTICE] mod_dingaling.c:715 Close Channel N/A [CS_NEW] 2010-10-13 12:15:41.432855 [DEBUG] switch_core_state_machine.c:430 () Running State Change CS_DESTROY 2010-10-13 12:15:41.432855 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY 2010-10-13 12:15:41.432855 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY going to sleep 2010-10-13 12:15:41.432855 [ERR] switch_ivr_originate.c:2605 Cannot create outgoing channel of type [dingaling] cause: [DESTINATION_OUT_OF_ORDER] 2010-10-13 12:15:41.432855 [DEBUG] switch_ivr_originate.c:3413 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2010-10-13 12:15:41.432855 [INFO] mod_dptools.c:2575 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2010-10-13 12:15:41.432855 [DEBUG] switch_channel.c:2445 (sofia/sipinterface_1/1001(a)192.168.1.200) Callstate Change RINGING -> HANGUP 2010-10-13 12:15:41.432855 [NOTICE] mod_dptools.c:2638 Hangup sofia/sipinterface_1/1001(a)192.168.1.200 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2010-10-13 12:15:41.432855 [DEBUG] switch_channel.c:2461 Send signal sofia/sipinterface_1/1001(a)192.168.1.200 [KILL] I think the interesting message is "Unknown Profile!" Here is the dialplan that is being matched: <extension name="dingaling_1_pattern_1" continue="true"> <condition field="destination_number" expression="^91{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$" bluebox="pattern_1"> <action application="set" bluebox="prepend" data="prepend=1"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="dingaling/GoogleTalk/+${prepend}$1(a)voice.google.com"/> </condition> </extension> Here is the profile: <profile name="dingaling_1"> <param name="name" value="GoogleTalk"/> <param name="login" value="justlikeef(a)gmail.com/gtalk"/> <param name="password" value="!Cl0r0x1.."/> <param name="server" value="talk.google.com"/> <param name="dialplan" value="XML"/> <param name="message" value=""/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="auto-login" value="true"/> <param name="auto-reply" value=""/> <param name="sasl" value="plain"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> <param name="context" value="context_1"/> <param name="exten" value="1001"/> </profile> Complete config and log in pastebin... -- Thanks, Rob

4 7

Problem with T.38 fax receive using ESL
by peely 30 Oct '10

30 Oct '10

Hi, I'm having problems receiving faxes through the Event Socket API using T.38 and SpanDSP. If I use the example dialplan entry: <extension name="NTS TESTfax"> <condition field="destination_number" expression="^\+(testfax)"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="rxfax" data="/tmp/faxes/FAX-${uuid}.tif"/> <action application="hangup"/> </condition> </extension> Then everything is OK and a fax is received nicely whereas if I use an ESL command on an answered call like: SendMsg 5716720c-e291-11df-939c-75e5ad3b2de7 call-command: execute execute-app-name: rxfax execute-app-arg: {fax_enable_t38_request=true,fax_enable_t38=true,fax_header=testheader,fax_ident=testident}/tmp/faxes/myfax.tif Then there's lots of T.38 mucking about but the fax terminates with a communications error. I've tried sleeping for 2 seconds, broadcasting silence_stream://2000 etc but to no avail. Has anyone successfully got T.38 fax receive working through the ESL, and if so, what chain of commands do you execute? I'm using a GIT trunk circa Oct 15th BTW. Many thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-T-38-fax-receive… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 4

wiki question - want to add Mac osX info
by Mario G 30 Oct '10

30 Oct '10

I am about to start work my wiki contribution which specifically target my installation on Mac osX and other related information to Mac osX. I saw there is no category for "installation". So then does this info go under "Examples"? Mario

2 1

Weekly Binaries for Windows are available
by Jeff Lenk 29 Oct '10

29 Oct '10

These evaluation binaries are built with VS2010(release mode) and are packaged in a MSI for easy distribution and installation. They are available in both X86 and X64 versions and install a base version of FreeSWITCH including all default modules along with 8Khz sounds. mod_managed can be used also but you will need to install the DotNet 4 Framework - http://www.microsoft.com/downloads/en/details.aspx?FamilyID=9cfb2d51-5ff4-4… Then enable the module in modules.conf. NOTICE!! Make sure if you install a newer version of the MSI you create backup copies of your /conf folder so you dont lose any configuration settings. New versions of the MSI will be available every week at approx 2am(GMT-6) Sunday morning barring any unforeseen problems. http://files.freeswitch.org/windows/installer/ The MSI will install v9 of the MSCRT libraries to your computer but otherwise makes no modifications other than placing files in your program files\FreeSWITCH folder and adding a "FreeSWITCH" menu entry in your start menu. This is a work in progress and will have frequent modifications if needed to correct any problems. You can also download and build the source code for yourself here. http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code Please enjoy and give feedback! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Weekly-Binaries-for-Windows-a… Sent from the freeswitch-users mailing list archive at Nabble.com.

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