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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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Re: [Freeswitch-users] Voicemail overriding leg_delay_start ? (more info)
by Paul Cupis 16 Jan '12

16 Jan '12

Try setting continue_on_fail to true before the bridge command. Regards,

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Bsuy, Wait, Callback - lua script..?
by Avi Marcus 15 Jan '12

15 Jan '12

Hi, I'm looking to implement a callback functionality. I get an incoming invite which has the user who is making the call and his destination. I'm supposed to 486 that and then after 2 second call them back, and connect the call. Should I do a hangup hook to a lua script which sleeps 2 seconds and does an originate? or zombie exec? And if I do the originate within the lua script, won't that leave the lua script running for the whole calls? What's the best way.. or even just a good way to do this - I don't expect more than 10 simultaneous calls as of now on this particular method. Thanks, -Avi

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Voicemail overriding leg_delay_start ? (more info)
by Bob Smith 15 Jan '12

15 Jan '12

Should add that I also tried the following syntax, in this instance, the second bridge number rings, but then on expiry, it never gets to voicemail, it just appears to drop: <action application="bind_meta_app" data="3 b s execute_extension::att_xfer XML features"/> <action application="set" data="domain_name=mydmain"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="{ignore_early_media=true} [leg_timeout=30]group/mygroup@mydmain,[leg_delay_start=12,leg_timeout=30]sofia/gateway/mygate/1234"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default mydmain 1000"/> </condition> </extension> </context> </section> Has anyone got this sort of setup running ?

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sched_hangup with playback
by Rodney 15 Jan '12

15 Jan '12

I would like to schedule a hangup at a set call time max but play a recording letting the person know the max time is up. is this possible in the dialplan? I have read http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup and see a reason code to set but no mention of playback warning the party. In addition, I would also like a 5 minute or so warning, like on a calling card platform. like "this call will end in ____ minutes" kind of thing. thanks.

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