September 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

how to append to a file using record
by covici＠ccs.covici.com 19 Sep '10

19 Sep '10

Hi. Is there anyway to append to a file using record? I see a variable record_append, but it seems to only effect record_session, or am I missing something here? 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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FreeSwitch on OSX server 10.6?
by Martin Joseph 18 Sep '10

18 Sep '10

Hello again y'all, I am curious as to whether any of you has built/used FreeSWITCH on OSX Server 10.6. I have recently purchased a used xserve (xeon 2.0G dually) on eBay, and I am considering finally making the switch from using FreeSWITCH as my backup (10.4 client on an old ibook G3) to my primary. Anyone done this? Thanks, Marty

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Building FS VS 2008 / For Windows Server
by Malay Thakershi 18 Sep '10

18 Sep '10

I started trying FS December last year and now I am in stage to deploy it in live environment with my projects. When I built it last year, I remember I had issues getting the configuration correct and ceptral issues. I do not have good administrative / config file skills. I would like to compile and deploy the new version to take advantage of bug fixes and new features such as google voice integration. My questions: 1. When I go to properties of the solution, I see certain items (such as 16Hz, json, etc) unchecked under "Configuration properties -- debug". Should I check all items to get a complete build? If I check everything I get build errors in JSON module. 2. What is purpose of "_Build system", "_Config" and "_Downloads" folders? Which config files are used to direct the solution to include certain things and exclude others? 3. When I did first install, I used windows setup to install FS. Should I overwrite the files that come out of solution building process on top of current folder? Is it possible to run multiple instances of FS on same computer so that I can check transition works? Thank you for any help or advice.

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FS stocked to 120ms in g723
by gamalsa 18 Sep '10

18 Sep '10

Hi I post an earlier topic about wrong ptime in g723. However after going into more troubleshooting, I found that FS uses only p-time 120 ms no matter what, even when calling internal extension or voice mail in FS using g.723 it uses always p-time 120 ms. The g.723 module loads fine with 30, 60, 90, and 120 ms, so I have no explanation of whats going wrong. Please Help.. Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-stocked-to-120ms-in-g723-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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dialing a alphanumeric user
by Adolfo Delorenzo 18 Sep '10

18 Sep '10

Hello, I am trying to call an alphanumeric user, for example john(a)domain.com and it doesn't work. I get a [NO_ROUTE_DESTINATION] error message. It only works if I add the number-alias option and dial the numeric extension. My user extension config (without number-alias) as follows: <include> <user id="john"> <params> <param name="password" value="1234"/> <param name="vm-password" value="1234"/> </params> <variables> <variable name="toll_allow" value=""/> <variable name="accountcode" value="john"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="John Doe"/> </variables> </user> </include> What am I doing wrong? Thanks

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Re: [Freeswitch-users] [?? Probable Spam] FreeSWITCH-users Digest, Vol 51, Issue 132
by sharad 18 Sep '10

18 Sep '10

Hi Is tt possible to record the voice of two IP phones who are registered with some other IPPBX & their voice is P2P. Means can FS record the rtp of non registered users ? Would really appreciate if someone could reply this. Regards Sharad ----- Original Message ----- From: <freeswitch-users-request(a)lists.freeswitch.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Saturday, September 18, 2010 1:24 PM Subject: [?? Probable Spam] FreeSWITCH-users Digest, Vol 51, Issue 132 > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." >

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sip stack problem in fsw
by covici＠ccs.covici.com 18 Sep '10

18 Sep '10

Hi. I am having a problem where -- to certain numbers, fsw hangs up after 5 minutes. Now this is suspiciously like a comfort noise problem, but I have no idea how to make fsw generate this, or can I? With asterisk, it does work using some providers, but fsw did not work with the same provider, so I am kind of stuck here. Thanks in advance for any suggestions. -- 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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Tips adding ASR to a existing script.
by broken dash 18 Sep '10

18 Sep '10

I'm wanting to add ASR to this javascript I've attached below... I've seen the pizza ordering demo as an example but I'm just not sure where to get started really an I'm looking for a little guidance. My script below has only 4 possible options (dtmf 1-4) and I'm simply just wanting to add the ability for a caller to speak the choice they would like ie: 1,2,3,or 4 while retaining the ability for the caller to enter the selection using dtmf. Any assistance would be appreciated... :-) Brian # cat autoattendant_5002.js var condition = true; var domain = "home"; //by default this is the ipv4 address of FreeSWITCH used for transfer to voicemail var digitmaxlength = 0; 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; 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", "ignore_early_media=true"); session.execute("set", "hangup_after_bridge=true"); session.execute("set", "continue_on_fail=true"); session.execute("set", "call_timeout=45"); session.execute("export", "call_timeout=45"); session.execute("set", "ringback="+us_ring); //set to ringtone session.execute("set", "transfer_ringback="+us_ring); //set to ringtone //console_log( "info", "Auto Attendant Server Time is: "+Hours+":"+Mins+" \n" ); function get_sofia_contact(extension,domain_name, profile){ if (profile == "auto") { profile = "external"; session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}"); sofia_contact = session.getVariable("sofia_contact_"+extension); if (sofia_contact == "error/user_not_registered") { profile = "internal"; session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}"); sofia_contact = session.getVariable("sofia_contact_"+extension); } } else { session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}"); sofia_contact = session.getVariable("sofia_contact_"+extension); } console_log( "info", "sofia_contact "+profile+": "+sofia_contact+".\n" ); return sofia_contact; } function mycb( session, type, obj, arg ) { try { if ( type == "dtmf" ) { console_log( "info", "digit: "+obj.digit+"\n" ); if ( obj.digit == "#" ) { //console_log( "info", "detected pound sign.\n" ); exit = true; return( false ); } dtmf.digits += obj.digit; if ( dtmf.digits.length >= digitmaxlength ) { exit = true; return( false ); } } } catch (e) { console_log( "err", e+"\n" ); } return( true ); } //end function mycb condition=true; if (condition) { //action //console_log( "info", "action call now don't wait for dtmf\n" ); var dtmf = new Object( ); dtmf.digits = ""; if ( session.ready( ) ) { session.answer( ); digitmaxlength = 1; while (session.ready() && ! exit ) { session.streamFile( "/usr/local/freeswitch/recordings/greeting-main.wav", mycb, "dtmf 20" ); if (session.ready()) { if (dtmf.digits.length == 0) { dtmf.digits += session.getDigits(1, "#", 20000); // 20 seconds if (dtmf.digits.length == 0) { } else { break; //dtmf found end the while loop } } } } //pickup the remaining digits dtmf.digits += session.getDigits(4, "#", 3000); //allow up to 5 digits if ( dtmf.digits.length > "0" ) { if ( dtmf.digits == "1" ) { // session.execute("transfer", "3973 XML default"); // } else if ( dtmf.digits == "2" ) { // session.execute("playback", "shout://stream.infowars.com"); // } else if ( dtmf.digits == "d" ) { // session.execute("bridge", "sofia/internal/5002@${domain}"); // } else if ( dtmf.digits == "3" ) { // session.execute("playback", "shout://ice.somafm.com/groovesalad"); // } else if ( dtmf.digits == "4" ) { // session.execute("bridge", "sofia/gateway/home/lisaworking"); // } else { //console_log( "info", "default option when there is no matching dtmf found\n" ); session.execute("bridge", "sofia/internal/5002@${domain}"); // } } } //end if session.ready }

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FreeSWITCH Conference Call Today - Discussing Cookbook Recipes, etc.
by Michael Collins 18 Sep '10

18 Sep '10

Okay all, here's the agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2010_09_15 Talk to you soon! -MC

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G723-Wrong ptime
by gamalsa 18 Sep '10

18 Sep '10

Hi I'm having a problem connecting to a provider using g723, FS read ptime as 120ms and compare it to ptime 20ms on the provider end, then chooses 120ms which end up with distorted voice. don't get any warning regarding wrong ptime reporting from other end. The provider said they are using ptime 30ms with g723, so have no idea why FS read it as 20 ms. g729 work great no problems. Is there a way to preset g723 ptime to 30 ms in FS. Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/G723-Wrong-ptime-tp5543130p55… Sent from the freeswitch-users mailing list archive at Nabble.com.

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