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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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New feature: sofia profile <profile> gwlist up|down
by DJB International 17 Sep '10

17 Sep '10

i wonder whether there is a way to use this command with a non-registered gateway. Thank you, Dorn.

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seeking consultant who can help (now) with SIP timeout issues
by Dale Trub 17 Sep '10

17 Sep '10

We're in Linux/FS and mysteriously calls are now dropping … any consultants out there who can decode with wireshark or whatever it takes, have real experience? Paid at full professional rates - but we need someone who is available today and if needed through this weekend to fix this. Please reply with contact info including phone number, rates, and something about your background. Thanks! -Dale

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wiki page about video conference set up in freeswitch
by jesse 17 Sep '10

17 Sep '10

do we have such wiki page? thanks!

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Hotword+MRCP
by Fernando Testa 17 Sep '10

17 Sep '10

Hi list, Do you know if hotword feature works with Nuance + FS + MRCPv2? If yes, how to accomplish that? Thank you -- Fernando Gregianin Testa Voice Technology Ltda

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A108 PCIe 2.0 support
by Anita Hall 17 Sep '10

17 Sep '10

Hi Does Sangoma 8-port card A108 support PCIe version 2.0 ? The card is here http://www.sangoma.com/products/hardware_products/digital_voice_and_data_ne… And we want to use 3 such cards in this motherboard because it has 3 PCIe slots of version 2.0 http://www.intel.com/products/desktop/motherboards/DX58SO/DX58SO-overview.h… Is this a good idea ? Do you have any experience with multiple A108 with PCIe on the same motherboard that supports PCIe 2.0 ? Any comments will be helpful. Thanks, Anita Hall Simmortel Voice.

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