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mod_h323 h323plus debian shlibdeps dependancy problem
by Steven Ayre 03 Nov '10

03 Nov '10

I have install ptlib 2.8.2 and h323plus trunk as per the instructions on the mod_h323 wiki page. When I build the debian packages it finishes with this error: dpkg-shlibdeps: failure: no dependency information found for /usr/local/lib/libh323_linux_x86_64_.so.1.22.0 (used by debian/freeswitch/opt/freeswitch/mod/mod_h323.so) Does anyone know what causes this error, and how to fix it? -Steve

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custom events
by Tihomir Culjaga 03 Nov '10

03 Nov '10

hello, im trying to get a custom event on an event socket outbound mode. on the eventsocket i subscribe only to "my events" with command 'myevents' and when i send a custom event, it seems not working ?!?! myevents Content-Type: command/reply Reply-Text: +OK Events Enabled event plain CUSTOM Content-Type: command/reply Reply-Text: +OK event listener enabled plain sendevent CUSTOM Event-Name: CUSTOM Event-Subclass: CUSTOM State: ACTIVE Content-Type: command/reply Reply-Text: +OK when i subscribe to all events ... im getting it correctly ... event plain all Content-Type: command/reply Reply-Text: +OK event listener enabled plain sendevent CUSTOM Event-Name: CUSTOM Event-Subclass: CUSTOM State: ACTIVE Content-Type: command/reply Reply-Text: +OK Content-Length: 469 Content-Type: text/event-plain Core-UUID: 04567a60-3b49-4645-82c8-4c5effd061a5 FreeSWITCH-Hostname: cc01 FreeSWITCH-IPv4: 89.201.165.245 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2010-11-03%2023%3A26%3A37 Event-Date-GMT: Wed,%2003%20Nov%202010%2022%3A26%3A37%20GMT Event-Date-Timestamp: 1288823197412378 Event-Calling-File: mod_event_socket.c Event-Calling-Function: read_packet Event-Calling-Line-Number: 1131 Command: sendevent%20CUSTOM%20 Event-Name: CUSTOM Event-Subclass: CUSTOM State: ACTIVE So, what the issue might be ? Whet m'i missing ?

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ISDN PRI Tapping
by lloyd thomas 03 Nov '10

03 Nov '10

Hi, I wish to use freeswitch to reocrd calls on a PRI ISDN span. I believe this may be pasiible with Sangoma boards. Are there any further instructions available than http://wiki.freeswitch.org/wiki/FreeTDM#Tapping ? Regards Lloydie-T

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Choosing the right distribution
by Bernhard Suttner 03 Nov '10

03 Nov '10

Hi, the time will come, when we will use FreeSWITCH on our systems. Currently we are using CentOS, but I am normally using debian systems because of the better software distribution, stability and update process. So? Are there any advantages if I would use CentOS or Debian? Does somebody has trouble on CentOS / Debian related to limitiation like Kernel configuration problems (I dont want to recompile the kernel for every system). Are there any analyses on which system FreeSWITCH does run "better" (stability, number of calls, performance,....) Any hint is appreciated. Best regards, Bernhard

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FreeSWITCH Conference Call Today - DRK Discusses .NET Fun
by Michael Collins 03 Nov '10

03 Nov '10

Hello all, The FreeSWITCH conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_11_03 Dave Kompel will be showing off some more .NET stuff that he's got cooking plus we have a few other things to discuss. Talk to you soon, Michael

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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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Dingaling/gtalk parking and bridging extension dialplan
by Majdi Bsoul 03 Nov '10

03 Nov '10

Hello guys, I am looking for your help and experience in putting a more efficient though hopefully simple extension dialplan to handle incoming Dingaling/gtalk call. This will help all Dingaling users especially with the limitation of Google Voice of having 25 seconds before forwarding the call to voicemail, since Google does not allow disabling the voicemail forwarding. Basically, I want to immediately allow FS to answer the Jingle/gtalk call and park it, but keep ringing the caller leg, then call the SIP extension and once it is answered to bridge the call with the parked Jingle/gtalk and stop ringing. The call flow: - Answer Jingle/gtalk - Park the call and keep ringing treatment back to caller - Call extension 1000 (SIP) - When 1000 answer => Pick up parked gtalk call and stop ringing. - If timeout (60 sec) => Disconnect both SIP and gtalk call legs (no voicemail forwarding) Thanks,

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Making calls without a gateway
by Kilburn Abrahams 03 Nov '10

03 Nov '10

Hi list I can make calls via a gateway, but am a bit stuck when making calls directly to sip provider. They accept IP authorization. I am using the following syntax. <action application="bridge" data="sofia/external/617$1@X.X.X.X:5060"/> I get [INFO] mod_dptools.c:2579 Originate Failed. Cause: NETWORK_OUT_OF_ORDER and on the phone I get BAD Gateway. When I use asterisk to the sip provider it works fine. Greatly appreciate any pointers. Regards Kilburn

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mod_xml_curl - Multitenant dialplan
by Aloysius Lloyd 02 Nov '10

02 Nov '10

Hi All, I would like to use mod_xml_curl in Multi-tenant environment. Also like to use dynamic and static files for dialplan. Here is the scenario *compa**.mydomain.com* Dial plan : *Dynamic *+ Staic File : * compa.mydomain.com.xml* *compb**.mydomain.com * Dial plan : *Dynamic *+ Staic File : * compb.mydomain.com.xml* -- Also find the following in wiki but that for single environment. <param name="dialplan" value="XML,XML:/path/to/custom/dialplan.xml"/> I could not find the correct way to implement this in Multi-tenant environment. Any help appreciated. Thanks Lloyd

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Fwd: Strategies for testing latency
by Darren Schreiber 02 Nov '10

02 Nov '10

Hi folks, This may be a slightly off-topic FreeSWITCH question, but here goes... I am having issues trying to figure out the source of latency in circuits we LCR to, it seems to be somewhat of a guessing game. I am embarking on creating a tool to measure this latency so we can test carriers more rapidly when deciding whether to use them or not. Before I re-invent the wheel, I'd love to know what others do to ensure latency is at a minimum on the circuits they buy from carriers OR if there are specific things to tweak in FreeSWITCH that I'm unaware of that somehow would help this. I can't imagine there are, but maybe I've missed something. It is my belief that some of the carriers we are using are behind Asterisk boxes that are always taking on media and are adding an encode/decode step to the path, resulting in some latency, as I can't find any evidence of late packets or high jitter on our box <-> carrier. For us, latency seems most noticable when we have callers on cell phones using a conference bridge. The latency nears 800-1000ms from when a person speaks to when they are heard on the other cell phone. I have tried to tweak our own conference bridge settings but I really don't think that's going to make a difference if my theory of the issue being upstream is correct. VoIP <-> VoIP conferencing has none of this issue. Any experience that folks can offer here is appreciated. Obviously getting a direct RTP stream from the carrier doing the conversion to PSTN would be best, but it has a tendency to require a lot of minutes or be expensive to do yourself. Thanks, Darren Schreiber Co-Founder - the 2600hz Project (415) 886-7900 - www.2600hz.org<http://www.2600hz.org/> Join us in January for FreeSWITCH Training in Australia! Visit www.voipkb.com<http://www.voipkb.com/> for more information

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