January 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

USER_NOT_REGISTERED with external profile
by Alexis de BRUYN [Mailinglists] 16 Jan '12

16 Jan '12

Hi Everybody, I am trying to use FreeSwitch in a (double) NAT Configuration from a fresh snapshot install (Debian Squeeze, server outside from my client LAN) with default directory for users. When I call 1001 from 1000 (or 1000 from 1001) , I cannot hear the callee, the phone doesn't ring and automatically hangup, I see in the console : 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] However, 1000 and 1001 are registered (from the same LAN) : sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.1.6 SIP-IP 192.168.1.6 URL sip:mod_sofia@192.168.1.6:5080 BIND-URL sip:mod_sofia@192.168.1.6:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 3 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: aU.1ZlEZkW7nlHpwp9ig1lt7A3weNnwm User: 1001(a)X.Y.Z.T Contact: "Freeswitch" <sip:1001@A.B.C.D:50193> Agent: Bria iOS 2.0.0 Status: Registered(UDP)(unknown) EXP(2012-01-16 23:16:00) EXPSECS(934) Host: phone IP: A.B.C.D Port: 50193 Auth-User: 1001 Auth-Realm: X.Y.Z.T MWI-Account: 1001(a)X.Y.Z.T Call-ID: 3c27b7424837-bsfi877ujb2n User: 1000(a)X.Y.Z.T Contact: "freeswitch" <sip:1000@192.168.0.69:2048;line=skgk4hug> Agent: snom300/8.4.32 Status: Registered(UDP)(unknown) EXP(2012-01-17 00:01:24) EXPSECS(3658) Host: phone IP: A.B.C.D Port: 62061 Auth-User: 1000 Auth-Realm: X.Y.Z.T MWI-Account: 1000(a)X.Y.Z.T Total items returned: 2 ================================================================================================= All necessary ports are opened/forwarded on the server. I See on the 1000 configuration that this is the local ip address which is set as contact. Is there any other setups to do in the directory ? Or other parameters in the external profile ? Thanks for your help ! Regards, -- Alexis de BRUYN Mail : alexis.mailinglist(a)de-bruyn.fr

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voicemail prompts are very quiet
by Ryan How 16 Jan '12

16 Jan '12

Hi, I've got a basic freeswitch install, just minor tweaks from the default (adding extensions, a gateway and DID). I find the volume of the voicemail prompts are very quiet. Recordings and recorded greetings are great. Is there a way to turn the volume of them up? I had a look through the wiki, but I can't seem to find anything with gain or volume. Maybe I am just looking for the wrong thing. Thanks, Ryan

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Re: [Freeswitch-users] Need Less CPU Intensive Way to Detect Voicemail and Answering Machines
by Bob Smith 16 Jan '12

16 Jan '12

How about something along these lines (http://blog.shimaore.net/2009/03/better-followme-for-freeswitch.html) If a user fails to confirm then assume it's a robot and move along your dialplan ? B

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Re: [Freeswitch-users] Javascript switch_ivr_originate.c:177 Invalid Application ?
by Bob Smith 16 Jan '12

16 Jan '12

Hi Avi, The pertinent lines are : <action application="set" data="group_confirm_key=exec"/> <action application="set" data="group_confirm_file=/usr/local/freeswitch/scripts/confirm.js"/> So I'm a step further now, I get: 2012-01-16 15:26:20.954602 [ERR] confirm.js:61 near SyntaxError: invalid returnreturn true; The same occurs if I use brackets around the "true". I've adapted the script slightly to use a pin pulled from a database (but I'm no Javascript Guru, so I'm guessing I missed a glaring bug !). Thanks for your help. ============================================================= // confirm.js - Freeswitch Call PIN Check // // console_log("info", "Destination: "+ session.destination + "\n"); // if(!session.getVariable('my_legconf')) { console_log("info", "No B-Leg Confirmation Requested \n"); exit(); } // // // Variables... // var v_continue = false; var conf_pin = session.getVariable('my_legpin'); var attempts = 3; var sound_file = "ivr-please_enter_pin_followed_by_pound.wav"; var pin_lastdigit=""; var pin = new Object(); pin.digits = ""; function parseInput (session,type,data,arg) { if ( type == "dtmf" ) { pin_lastdigit = data.digit; pin.digits += pin_lastdigit; console.log("info","Read digit " + pin_lastdigit + "\n"); return false; } } // // if (session.ready()) { session.answer(); session.flushDigits(); console.log("info","Starting PIN Collection\n"); var cnt = attempts; while (session.ready() && ! v_continue && cnt-- > 0) { session.execute("sleep","200"); session.streamFile(sound_file,parseInput); while (pin_lastdigit != "#" && sesson.ready()) { session.collectInput(parseInput,pin,5000); } console.log("info","Collected PIN: " + pin.digits + "\n"); if ( pin.digits == conf_pin ) { v_continue = true; console.log("info","PIN OK !.\n"); } else { pin_lastdigit = ""; pin.digits = ""; session.flushDigits(); } } } else { console.log("info","Session not ready.\n"); } // // if(v_continue) { return true; } else { return false; } =============================================================

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Javascript switch_ivr_originate.c:177 Invalid Application ?
by Bob Smith 16 Jan '12

16 Jan '12

Hi, I'm trying to get an authenticated bridge going along the lines of the one featured on this website (http://blog.shimaore.net/2009/03/better-followme-for-freeswitch.html) However I'm getting the following error in the console : 2012-01-16 14:52:43.174602 [ERR] switch_ivr_originate.c:177 Invalid Application /usr/local/freeswitch/scripts/confirm.js Is there any special magic required that the blog poster forgot to mention ? (e.g. some includes or similar in the Javascript ?). Bob

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Re: [Freeswitch-users] Voicemail overriding leg_delay_start ?
by Bob Smith 16 Jan '12

16 Jan '12

Thanks Paul. Will try. > Try setting continue_on_fail to true before the bridge command

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Caller-Caller-ID-Number on originate
by Tux Tux 16 Jan '12

16 Jan '12

Hi, When trying this: originate {sip_callee_id_numberorigination_caller_id_name=103,origination_caller_id_number=103,instant_ringback=true,ignore_early_media=true,ringback=\'%(1000,4000,425.0,0.0)\'}user/102 &transfer('103 XML default') Caller-Caller-ID-Number will eventually be set to: internal/sip:102@192.168.57.209:32710;rinstance=44018f313d6dd1aa While when using two softphones, the Caller-Caller-ID-Number will stay 102 instead of internal/sip:102@192.168.57.209:32710;rinstance=44018f313d6dd1aa Can I set a channel variable to keep the Caller-Caller-ID-Number the same? Discard my last email please. Thanks, Nico

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Caller-Caller-ID-Number on originate using ESL
by Tux Tux 16 Jan '12

16 Jan '12

originate {sip_callee_id_numberorigination_caller_id_name=103,origination_caller_id_number=103,instant_ringback=true,ignore_early_media=true,ringback=\'%(1000,4000,425.0,0.0)\'}user/102 &transfer('103 XML amteam')

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cdr minutes of use per did
by Rodney 16 Jan '12

16 Jan '12

Avi, thank you for quick response, I am using sqlite with fusionpbx but it is in windows, so I don't believe I can run cron jobs in windows. ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Saturday, January 14, 2012 6:03 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 137 ----- Forwarded Message ----- 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..." Today's Topics: 1. cdr minutes of use per did (Rodney) 2. Re: Problems with CURL_XML and bridging ? (Bob Smith) 3. Re: cdr minutes of use per did (Avi Marcus) does anyone have a cdr viewer or method that can help me determine the total minutes for all calls on a specified DID for the hour/day/month to date? I would like a method to generate such a report and maybe have it emailed on a 24 hour basis to the person in charge of the DID. the following I have seen in the past with any method to find the original program that would give an output like this. Calls by Day, DNIS -> DID ACCOUNT-> Friday, 8/19/11 (Eastern Standard Time) Daily Month to Date (Start of Month = Day 21) Day DNIS DNIS Group Total Calls Minutes of Use Average Hold Time (Minutes) Total Calls Minutes of Use Average Hold Time (Minutes) – No rows – Actually, you're quite right. I've gone back and double checked. Does not work with static XML. Same behaviour... transfer works, bridge fails. However I can definitely originate a call from the CLI over the same route ..... "originate sofia/gateway/mygate/12345678 &park" works great. That's what's really bugging me, if I can CLI originate to that number, if I can call that PSTN number from my FreeSwitch softphone, why on earth can't i do this simple bridge ! ;-( >So doing the bridge to the same number with static XML worked, but when > bridging to the exact same endpoint with xml_curl you're getting a 503? It depends on how your cdr statistics are stored.. CSV file, sqlite, pgsql, mysql.. for how easy it is to run a query. And if you want just inbound or outbound "on" that DID. Then just add that SQL query to a script and have a cron job run it.. e.g. if you use mysql in fusionpbx to store your cdrs (via xml_cdr posts) I can run: SELECT SUM(`billsec`)/60 FROM `v_xml_cdr` WHERE `sip_to_user` = '$number' AND `start_stamp` LIKE '2011-11%' = 831.4500 minutes for all of November 2011. (pgsql doesn't like using a like like that on a time stamp field, it seems..) Your query, and source info, will vary.. -Avi On Sun, Jan 15, 2012 at 12:41 AM, Rodney <notlikeme75(a)yahoo.com> wrote: does anyone have a cdr viewer or method that can help me determine the total minutes for all calls on a specified DID for the hour/day/month to date? I would like a method to generate such a report and maybe have it emailed on a 24 hour basis to the person in charge of the DID. the following I have seen in the past with any method to find the original program that would give an output like this. >Calls by Day, DNIS -> DID ACCOUNT-> Friday, 8/19/11 (Eastern Standard Time) > >Daily >Month to Date >(Start of Month = Day 21) >Day >DNIS >DNIS >Group >Total >Calls >Minutes >of Use >Average >Hold Time >(Minutes) >Total >Calls >Minutes >of Use >Average >Hold Time >(Minutes) >– No rows – >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Effect of sync_clock on calls?
by Michael Giagnocavo 16 Jan '12

16 Jan '12

Should there be any audio-quality effects on calls by running fsctl sync_clock? To try to get my timestamps as close as possible to system time, I plan on running this every 5 minutes or so. Is there a fundamental issue with this? Or is there a way to disable the monotonic clock completely? I understand what it's trying to achieve, but I cannot see how it'll ever be any more accurate than the system time, which is sync'd via NTP. I'm more concerned about my timestamps being accurate than protecting against someone turning off NTP and changing the time to some invalid value. -Michael

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