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recvEventTimed - SERVER_DISCONNECTED
by Nagalenoj H. 09 Mar '10

09 Mar '10

Dear friends, I've faced an issue in event socket. I would want to know why it behaves such a way. My program is working with the hep of events. So, based on the received event, the process will continue it's work. When I need DTMF, I use recvEventTimed and in the other cases, I use recvEvent. So, In the mid if caller hangsup, I expect for the SERVER_DISCONNECTED event. When the caller hangsup when I'm waiting in recvEvent, I'm getting SERVER_DISCONNECTED. But, when I'm waiting in recvEventTimed, I'mnot receiving SERVER_DISCONNECTED, instead receiving an undefined value. To handle this, I've checked esl connection inside the timeout part and put a recvEvent, then I receive SERVER_DISCONNECTED. My question is, why am I not receiving SERVER_DISCONNECTED when I recv event using recvEventTimed?! This is a sample program I execute, require ESL; use IO::Socket::INET; my $ip = "127.0.0.1"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '9242', Proto => 'tcp', Listen => 1, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; for(;;) { my $new_sock = $sock->accept(); my $pid = fork(); if ($pid) { close($new_sock); next; } my $host = $new_sock->sockhost(); my $fd = fileno($new_sock); 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"); my $e = $con->filter("unique-id", $uuid); if ($e) { print $e->serialize(); } else { printf("WTF?\n"); } $con->events("plain", "SERVER_DISCONNECTED DTMF"); $con->execute("answer"); while($con->connected()) { $e = $con->recvEvent(); ############# CASE 1 ## # $e = $con->recvEventTimed(10000); ############# CASE 2 ## unless ($e) { unless ($con->connected()) { ############################### $e = $con->recvEvent(); ## print $e->serialize(); ## Added code to get SERVER_DISCONNECTED. print "SERVER_DISCONNECTED"; ## } ############################### print "DTMF timeout\n"; } if ($e) { my $name = $e->getHeader("event-name"); print "EVENT [$name]\n"; if ($name eq "DTMF") { my $digit = $e->getHeader("dtmf-digit"); my $duration = $e->getHeader("dtmf-duration"); print "DTMF digit $digit ($duration)\n"; } } } print "BYE\n"; close($new_sock); } -- Regards, Nagalenoj H.

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create phone user extension
by Madovsky 09 Mar '10

09 Mar '10

to create a phone user extension, is it need to have his account in conf/directory/default for default.xml ? if yes so I don't understand why I can register with extension 1003 if there are only 1000-1002 in conf/directory I use the defaults FS config Thanks

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Which Lua script am I?
by Mike van Lammeren 09 Mar '10

09 Mar '10

Hello! I am working on a high-availability system and need a Lua script that can run in the background and loop forever, periodically waking up to accomplish some tasks. I would like to be able to handle at least 10 concurrent tasks. My first idea was to have a simple background script that would call another script, let's call it the 'task' script, every 5 seconds or so. This would spread the work around, and if any task script quit with an error, the background script would remain running, and would continue to launch new task scripts. However, I can see no way for the background script to launch other Lua scripts. Am I missing something? I need something like session:execute(), but unfortunately this background script has no session, and I'm not sure if session:execute() can even make a call like the "luarun" console command. So, my next ideas was to have 10 copies of the background script. Each script would loop forever. Upon launch, each script would perform a single task, then sleep for 60 seconds. This allows handling of 10 concurrent tasks. However, there are some problems with running 10 copies of the same script, which I believe can be solved if each script could be given a unique identifier. One problem is that I would like the scripts to actually sleep for a variable amount of time, say between 55 and 65 seconds, so that the workload would spread around. Another, similar problem, is that each script needs to 'claim' a number of database records from a table that is queuing up work for the scripts. Once again, if each script had a unique identifier, then I could use an update query to effectively 'set and test' a record that would be tied to that script instance. I could use a random number to identify each script, but since math.random is seeded with a timestamp that only resolves to seconds, then all 10 scripts will be given the same sequence of random numbers, and all 10 will run in lock-step with each other. So, random numbers are out. (If I had a unique ID to seed the random number generator, then I could just use that unique ID and wouldn't need random numbers.) Is there a way of uniquely identifying 10 Lua scripts that FreeSWITCH launches at startup? Or is there a better way of handling this situation altogether? I'm open to any suggestions that people would care to make! (I would entertain the idea of a single script that would start with FreeSWITCH, then launch 10 other scripts, giving each a unique identifier, but that brings me back to the original problem of not knowing how to launch other Lua scripts under FreeSWITCH.) Mike van Lammeren

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hunt group - play music when trying external numbers
by William Traenkle 09 Mar '10

09 Mar '10

I did not see this message go through yet so I removed the code I originally had pasted and am resending it. FS Community, I am new to freeSWITCH and this mailing list and I appreciate your support in advance. My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried. The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried. Ideas? Please advise. Thanks. -Will

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mod_spidermonkey
by Matt Putnam 08 Mar '10

08 Mar '10

http://jira.freeswitch.org/browse/MODLANG-157?page=com.atlassian.jira.plugi… Has anyone found a solution to the above issue? I am having the same issue on CentOS 5.4 with freeswitch version 1.0.5-20100303-0400. Thanks, Matt

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CID Question: How pass the CID?
by Javier Aristizábal 08 Mar '10

08 Mar '10

Hi folks!! I have an incoming call being bridged to an outgoing call ann i'm trying to pass the phone number of the incoming caller as the caller id on the leg B. But it doesn't work. Here is my dialplan: <extension name="incoming"> <condition field="destination_number" expression="^(3055908934)$"> <action application="info"/> <action application="set" data="effective_callee_id_number=${caller_id_number}"/> <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/xcast/17862065658"/> </condition> </extension> Here is the Log and SIP trace: http://pastebin.freeswitch.org/12368 This is the INVITE on the bLeg: INVITE sip:17862065658@wholesale.xcastlabs.com<sip%3A17862065658(a)wholesale.xcastlabs.com>SIP/2.0 Via: SIP/2.0/UDP 66.231.242.173:5080;rport;branch=z9hG4bKS6jFBtp9crtFp Max-Forwards: 68 From: "Javier" <sip:xcast@wholesale.xcastlabs.com<sip%3Axcast(a)wholesale.xcastlabs.com> ;transport=udp>;tag=K6U13Hy2rpgBe To: <sip:17862065658@wholesale.xcastlabs.com<sip%3A17862065658(a)wholesale.xcastlabs.com> > Call-ID: 0c42bf79-a59c-122d-b98c-0014384f7ec2 CSeq: 127923447 INVITE Contact: <sip:gw+xcast@66.231.242.173:5080;transport=udp;gw=xcast> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16921 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 250 X-FS-Support: update_display Remote-Party-ID: "Javier" <sip:3052606062@wholesale.xcastlabs.com<sip%3A3052606062(a)wholesale.xcastlabs.com> >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1268060050 1268060051 IN IP4 66.231.242.173 s=FreeSWITCH c=IN IP4 66.231.242.173 t=0 0 m=audio 23516 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 And i'm on FreeSWITCH Version 1.0.trunk (16921) Many thanks /Javier

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hunt group - play music when trying external numbers
by William Traenkle 08 Mar '10

08 Mar '10

I am sending this again as I did not see it go through the first time ... FS Community, I am new to freeSWITCH and this mailing list and I appreciate your support in advance. My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried. The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried. Ideas? Please advise. My hunt group code is below: session.answer(); var domain_name = session.getVariable("domain_name"); var extension = '7000'; var result; var timeoutpin = 7500; var objdate = new Date(); var adjusthours = 0; //Adjust Server time that is set to GMT 7 hours var adjustoperator = "-"; //+ or - if (adjustoperator == "-") { var objdate2 = new Date(objdate.getFullYear(),objdate.getMonth(),objdate.getDate(),(objdate.getHours() - adjusthours),objdate.getMinutes(),objdate.getSeconds()); } if (adjustoperator == "+") { var objdate2 = new Date(objdate.getFullYear(),objdate.getMonth(),objdate.getDate(),(objdate.getHours() + adjusthours),objdate.getMinutes(),objdate.getSeconds()); } var Hours = objdate2.getHours(); var Mins = objdate2.getMinutes(); var Seconds = objdate2.getSeconds(); var Month = objdate2.getMonth() + 1; var Date = objdate2.getDate(); var Year = objdate2.getYear() var Day = objdate2.getDay()+1; var exit = false; function get_sofia_contact(extension,domain_name, profile){ if (profile == "auto") { profile = "internal"; session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}"); sofia_contact = session.getVariable("sofia_contact_"+extension); if (sofia_contact == "error/user_not_registered") { profile = "external"; session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}"); sofia_contact = session.getVariable("sofia_contact_"+extension); } } else { session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}"); sofia_contact = session.getVariable("sofia_contact_"+extension); } console_log( "info", "sofia_contact "+profile+": "+sofia_contact+".\n" ); return sofia_contact; } function mycb( session, type, obj, arg ) { try { if ( type == "dtmf" ) { console_log( "info", "digit: "+obj.digit+"\n" ); if ( obj.digit == "#" ) { //console_log( "info", "detected pound sign.\n" ); exit = true; return( false ); } dtmf.digits += obj.digit; if ( dtmf.digits.length >= digitmaxlength ) { exit = true; return( false ); } } } catch (e) { console_log( "err", e+"\n" ); } return( true ); } //end function mycb dialed_extension = session.getVariable("dialed_extension"); domain_name = session.getVariable("domain_name"); domain = session.getVariable("domain"); us_ring = session.getVariable("us-ring"); caller_id_name = session.getVariable("caller_id_name"); caller_id_number = session.getVariable("caller_id_number"); effective_caller_id_name = session.getVariable("effective_caller_id_name"); effective_caller_id_number = session.getVariable("effective_caller_id_number"); outbound_caller_id_name = session.getVariable("outbound_caller_id_name"); outbound_caller_id_number = session.getVariable("outbound_caller_id_number"); session.execute("set", "ringback=${hold_music}"); //set to ringtone session.execute("set", "transfer_ringback=${hold_music}"); //set to ringtone session.execute("set", "call_timeout=30"); session.execute("export", "call_timeout=30"); session.execute("set", "continue_on_fail=true"); session.execute("set", "hangup_after_bridge=true"); //console_log( "info", "dialed extension:"+dialed_extension+".\n" ); //console_log( "info", "domain: "+domain+".\n" ); //console_log( "info", "us_ring: "+us_ring+".\n" ); //console_log( "info", "domain: "+domain+".\n" ); //console_log( "info", "domain_name: "+domain_name+".\n" ); //console_log( "info", "action call now don't wait for dtmf\n" ); if ( session.ready() ) { session.answer(); sofia_contact_1000 = get_sofia_contact("1000",domain_name, "auto"); session.execute("bridge", sofia_contact_1000); session.execute("bridge", "sofia/gateway/flowroute.com/16194548924"); //timeout if (session.getVariable("originate_disposition")!="SUCCESS" && session.getVariable("originate_disposition")!="ORIGINATOR_CANCEL"){ session.execute("info", ""); session.execute("transfer", "*991000"); } //clear variables dialed_extension = ""; new_extension = ""; domain_name = ""; domain = ""; } //end if session.ready Thanks. -Will

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g711 to g722 transcoding causing distortion on Polycom IP6000 handsets
by Dan Lane 08 Mar '10

08 Mar '10

This is almost certainly more of a Polycom issue than a FreeSWITCH one but I'll mention it here on the offchance that someone else has experienced the following: Loud G.711a calls that are being transcoded to G.722 are massively distorted on Polycom IP6000 handsets and on all other handsets the call just sounds loud and isn't distorted. This is most common in conference calls where callers are using a mixture of codecs. If I disable G.722 the Polycom sounds fine even though the G.711a call is still too loud and if I lower the volume of the conference the distortion still happens. So, has anyone else experienced this and if so, did you find a solution? Regards, Dan Lane

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XML curl and multiple extension in dialplan
by paul gore 08 Mar '10

08 Mar '10

Greetings, We use XML curl to fetch our dial plan extensions from a database. As per FS wiki we can only return one XML extension upon FS request. Now we need to return a set of extensions with same number condition and context but with different second condition so that FS can pick up the right one. Is it possible at all? And if yes how response XML with multiple extensions should look like? Thanks a lot.

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Send a call directly to voicemail Transfer Help
by Aloysius Lloyd 08 Mar '10

08 Mar '10

Hi All, I would like to setup a dial plan, Dial *99[extension number] send a call directly to voicemail. I setup the following Dial Plan . But when I transfer the call *99[extension number] the call failed. <extension name="voicemail_transfer"> <condition field="destination_number" expression="^(\*9920[0-9])$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> But when I transfer the call *99[extension number] the call failed.My extensions range 201 - 209 Please let me know where the mistake is. Thanks Lloyd

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