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Multiple Registrations - randomly one device ring
by Aloysius Lloyd 10 Mar '10

10 Mar '10

Hi All, I enable the multiple registrations in sip_profiles/internal.xml. <param name="multiple-registrations" value="true"/> Then I restart the freeswitch. I use Aastra 9133i and eyebeam register for same extension. But randomly only one device ring. Do I need to enable any other settings? Thanks, Lloyd

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FreeSWITCH Conference Call Agenda
by Michael Collins 10 Mar '10

10 Mar '10

Hello all! The FreeSWITCH weekly conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_03_10 Please feel free to add your questions and agenda items. Also, remember that this week Mathieu Rene is going to be spending a lot of time discussing mod_sofia and how it works. It should be a very informative discussion! By all means join us at 1700 GMT on Wednesday. -Michael

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

10 Mar '10

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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IVR Menu - Timeout and Invalid Options
by Aloysius Lloyd 10 Mar '10

10 Mar '10

Hi All, I am trying to setup a simple IVR with two options , timeout action and Invalid digits pressed. here is the partial IVR Menu. <entry action="menu-exec-app" digits="1" param="execute_extension 201 XML default"/> <entry action="menu-exec-app" digits="2" param="execute_extension 202 XML default"/> How to Catch Timeout and Invalid Digits Actions from the IVR menu? Thanks you Thanks, Lloyd

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Re: [Freeswitch-users] vars from sip_registrations table
by Madovsky 10 Mar '10

10 Mar '10

forgot to say that I need it (since I didn't find any good examples) for a 2 FS clustered without SRV NAPTR in DNS (the client I should use is not compatible) Thanks F ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 09, 2010 2:59 AM Subject: vars from sip_registrations table in sip_registration table there are call_id sip_user sip_host presence_hosts contact status rpid expires user_agent server_user server_host profile_name hostname network_ip network_port sip_username sip_realm mwi_user mwi_host orig_server_host orig_hostname fields, so how can I get these vars from the callee before the bridge ? I need to replace the host of the callee in case of if he's registered on another FS node. thanks Franck

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How to originate a new call from mod_managed?
by Chris 10 Mar '10

10 Mar '10

Hi, I am trying to create a mod_managed API application that takes 2 phone numbers as params. These numbers should then be used to make 2 calls and to bridge them. The issue i am having is working out how to place the first call leg from in mod_managed. All the methods i have found are for the second leg and require you to pass in the first call leg. I know there are easier way to do this out of the managed code but i want it in the managed code as this is just a proof of concept to prove we can do a callback like system via managed code to be hooked into other external systems. In wiki http://wiki.freeswitch.org/wiki/Session there is this example s = new Session("{ignore_early_media=true}sofia/default/foo(a)bar.com"); while (s.ready()) { // The call has been answered } This seems to be exactly what i am looking for but seems to be missing from mod_managed. I am hoping someone can tell me how to do something similar in mod_managed since even if it is not part of the managed code mod_managed is meant to have the native access as well so i would assume it would be possible. Thanks in advance Chris

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call routing from freeswitch based on INVITE
by srinivasula reddy 10 Mar '10

10 Mar '10

I have two sipservers like server1 and server2, if sever1 receives invite packet like INVITE From: 1000(a)server1.domain.com To: 1002(a)server2.railvoice.com. how can i route the invite packet to server2 from server1, Thanks & Regards Srinivasula Reddy K

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How to tie context to a gateway?
by Nikolay Kondratyev 10 Mar '10

10 Mar '10

Hi all, I have several gateways in the external profile. Let's say GW1 and GW2. I'd like to process calls from the GW1 in the context C1 and calls from GW2 in the context C2. Parameter "context", as far as I understand works for the whole profile, not for individual gateways in the profile. How do send calls from GW1 into context C1? What will be a good practice to do that? Thanks in advance, Nikolay.

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actition after a set time during call
by Todd 10 Mar '10

10 Mar '10

Hey List- I want to have nibblebill pause after a certain time during a call. I was wondering what the best way to put this into the dialplan is? Still kind of new to this.. <action application="nibblebill" data="pause"/> Is the action I want to implement 2 minutes into a call. what is the best way to do this? I have the nibblerate set in the individual extension XMLs and the nibblerate heartbeat set in the nibble.conf.xml Thanks

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freeswitch to cisco router
by srinivasula reddy 09 Mar '10

09 Mar '10

Hi, i have tried to configure gateway from my freeswitch, with the following credentials proxy(ip of cisco) and username,password. and when i start my freeswitch registration reqest is going to cisco router and again it requesting to same registration reqeust to my sipserver, the register packet from cisco router to my freeswitch is comming like this To: sip:.*@sipserverip:5060 so there there is no user present with the name of .* it is rejecting the registration request, any idea Thanks Swapna

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