July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Voice to email
by Karl Schmidt 27 Jul '13

27 Jul '13

I see the voicemail app - but I want to do something much simpler and reading through the books and docs there seems to be several ways to approach this. All I want to do is set up an extension to send unanswered calls to it will: - Play a recording (please leave a message.. etc) - Record the message - hangup - then email the message and not leave it behind in a voicemail box. I can figure out the details, but my question is the BEST way to do this - a script or messing with the voicemail app? (IMO this is a much better way to handle voice-mail - you check for messages at the same time you check your email - and listening to voicemail in vlc is much better than over a phone - you can scrub back to figure out the number/address etc.. You can forward the message via email if you need to.) -------------------------------------------------------------------------------- Karl Schmidt EMail Karl(a)xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 Prosperity is the best protector of principle. -- Mark Twain --------------------------------------------------------------------------------

6 11

Issues with using avmd from python script
by Raghu 27 Jul '13

27 Jul '13

I am new to freeswitch. I am trying to send a message to voicemail using avmd for voicemail detection, here is the code which i am using with originate command: from freeswitch import * # WARNING: known bugs with hangup hooks, use with extreme caution def hangup_hook(session, what): consoleLog("info","hangup hook %s!!\n\n" % what) return def input_callback(session, what, obj): if (what == "dtmf"): consoleLog("info", what + " from callback " + obj.digit + "\n") return elif (what == "event" and session.getVariable('avmd_detect') == "TRUE"): consoleLog("info", "Voicemail Detected\n") return "true" def handler(session, args): session.answer() session.execute("avmd", "start") session.execute("sleep", "25000") callback = session.setInputCallback(input_callback) consoleLog("info","callback returned %s!!\n\n" % callback) while session.getVariable('avmd_detect'): session.execute("speak", "flite|kal|'Hi how are you'") session.execute("avmd", "stop") session.setHangupHook(hangup_hook) session.hangup() #hangup the call I can see avmd getting started but it does detect any beep. Is there a simpler way to check if avmd is working without using 'originate'? Thanks

3 12

call-id or from variable?
by Paul 26 Jul '13

26 Jul '13

Hi guys, I'm sure this is an easy one for an experienced freeswitch user. I have the following scenario for outgoing calls: FS-Box2 --> FS-Box1 --> SIP Provider Currently my call ID is always set to the username/pilot # of the gateway parameter on the FS-Box1 which sets up the trunk to the sip provider. Question is, how do I pass the proper call-id number that's set on FS-Box2 through Box 1 and then out to the world? I have FS-Box2 registering to FS-Box1 as a user (in conf/directory/default/DIDNUM) in there I have tried to set: <param name="from-user" value="DID_NUM"/> (not literally DID_NUM) variable name="effective_caller_id_name" "effective_caller_id_number" "outbound_caller_id_name" "outbound_caller_id_number" even tried "outbound_from" value="DID_NUM" no success :) if anyone can see my folly I would be appreciative! Thanks Paul

2 1

Script for resubmitting failed mod_xml_cdr CDRs
by Steven Ayre 26 Jul '13

26 Jul '13

Hi everyone, For anyone using mod_xml_cdr to submit CDRs to a web server, you'll be familiar with the issue that if there is a web server error the CDRs will not be loaded into the database. They need to be manually resubmitted when the error is corrected to appear in the DB. I have added a script to the Wiki that can automatically find and resubmit these CDRs. You could either use this from the command-line or schedule it in a cronjob. https://wiki.freeswitch.org/wiki/Mod_xml_cdr#Script_.28for_cronjob_or_manua… It will automatically remove the failed CDRs from the err-log-dir, but only if the HTTP request is successful. It requires minimum configuration, as it reads the required url & credentials from the existing mod_xml_cdr.conf.xml file. -Steve

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Re: [Freeswitch-users] Issues with using avmd from python script
by raj singh 26 Jul '13

26 Jul '13

I have changed my email , not sure if you got my last email so sending again, thanks for your help On Fri, Jul 26, 2013 at 11:48 AM, Rajput R Singh <srraghu45(a)gmail.com>wrote: > > > Begin forwarded message: > > *From: *Michael Collins <msc(a)freeswitch.org> > *Subject: **Re: [Freeswitch-users] Issues with using avmd from python > script* > *Date: *July 25, 2013 12:14:49 PM PDT > *To: *FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Reply-To: *FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > I copied the logs to http://pastebin.freeswitch.org/21239 so that I could > see them more easily. It looks to me like you're just hanging up the call > after you detect the beep. If you want to do something else I would just > add whatever dialplan apps you want to run. > > -MC > > > On Thu, Jul 25, 2013 at 10:56 AM, Raghu <srraghu45(a)gmail.com> wrote: > >> I have pasted the console debug logs here <http://pastebin.com/anN768su> , >> please review it >> >> Thanks >> >> >> On Thu, Jul 25, 2013 at 9:06 AM, Michael Collins <msc(a)freeswitch.org>wrote: >> >>> Get a complete console debug log of the call and put onto pastebin so >>> that the gang here can review what's happening. >>> -MC >>> >>> >>> On Wed, Jul 24, 2013 at 11:01 PM, Raghu <srraghu45(a)gmail.com> wrote: >>> >>>> Thanks MC, i made the change and avmd was able to detect beep on few >>>> occasions everything else remaining same. Also, i wasnt able to send the >>>> message to voicemail, i tried with invoking sleep of 3secs >>>> >>>> ---------------------------------------- >>>> from freeswitch import * >>>> >>>> # WARNING: known bugs with hangup hooks, use with extreme caution >>>> def hangup_hook(session, what): >>>> >>>> consoleLog("info","hangup hook for RAJ %s!!\n\n" % what) >>>> return >>>> >>>> def input_callback(session, what, obj): >>>> >>>> if (what == "dtmf"): >>>> consoleLog("info", what + " from callback " + obj.digit + "\n") >>>> return >>>> elif (what == "event" and session.getVariable('avmd_detect') == >>>> "TRUE"): >>>> consoleLog("info", "Voicemail Detected\n") >>>> >>>> >>>> return "TRUE" >>>> >>>> def handler(session, args): >>>> >>>> session.answer() >>>> callback = session.setInputCallback("input_callback") >>>> session.execute("avmd", "start") >>>> consoleLog("info", "AVMD start\n") >>>> session.execute("sleep", "16000") >>>> consoleLog("info", "AVMD status %s\n" % >>>> session.getVariable('avmd_detect')) >>>> if session.getVariable('avmd_detect'): >>>> session.execute("sleep", "3000") >>>> consoleLog("info", "Beep Detected\n") >>>> session.execute("speak", "flite|kal|'Hi this is '") >>>> session.execute("avmd", "stop") >>>> session.execute("speak", "flite|kal|'Hi this is '") >>>> consoleLog("info","callback returned for %s!!\n\n" % callback) >>>> session.execute("avmd", "stop") >>>> consoleLog("info", "AVMD stop\n") >>>> session.setHangupHook(hangup_hook) >>>> >>>> >>>> session.hangup() #hangup the call >>>> ----------------------------------------------------------------- >>>> >>>> Please suggest >>>> >>>> Thanks >>>> >>>> >>>> On Wed, Jul 24, 2013 at 4:08 PM, Michael Collins <msc(a)freeswitch.org>wrote: >>>> >>>>> Hi Raghu, >>>>> >>>>> I would put the setInputCallback before the sleep, that way if the >>>>> avmd is detected while you're sleeping then it should work. >>>>> >>>>> -MC >>>>> >>>>> >>>>> On Wed, Jul 24, 2013 at 1:30 PM, Raghu <srraghu45(a)gmail.com> wrote: >>>>> >>>>>> I am new to freeswitch. I am trying to send a message to voicemail >>>>>> using avmd for voicemail detection, here is the code which i am using with >>>>>> originate command: >>>>>> >>>>>> from freeswitch import * >>>>>> >>>>>> # WARNING: known bugs with hangup hooks, use with extreme caution >>>>>> def hangup_hook(session, what): >>>>>> >>>>>> consoleLog("info","hangup hook %s!!\n\n" % what) >>>>>> return >>>>>> >>>>>> def input_callback(session, what, obj): >>>>>> >>>>>> if (what == "dtmf"): >>>>>> consoleLog("info", what + " from callback " + obj.digit + >>>>>> "\n") >>>>>> return >>>>>> elif (what == "event" and session.getVariable('avmd_detect') == >>>>>> "TRUE"): >>>>>> consoleLog("info", "Voicemail Detected\n") >>>>>> return "true" >>>>>> >>>>>> def handler(session, args): >>>>>> >>>>>> session.answer() >>>>>> session.execute("avmd", "start") >>>>>> session.execute("sleep", "25000") >>>>>> callback = session.setInputCallback(input_callback) >>>>>> consoleLog("info","callback returned %s!!\n\n" % callback) >>>>>> while session.getVariable('avmd_detect'): >>>>>> session.execute("speak", "flite|kal|'Hi how are you'") >>>>>> >>>>>> session.execute("avmd", "stop") >>>>>> session.setHangupHook(hangup_hook) >>>>>> session.hangup() #hangup the call >>>>>> >>>>>> >>>>>> I can see avmd getting started but it does detect any beep. Is there >>>>>> a simpler way to check if avmd is working without using 'originate'? >>>>>> >>>>>> Thanks >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> 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 >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Michael S Collins >>>>> Twitter: @mercutioviz >>>>> http://www.FreeSWITCH.org <http://www.freeswitch.org/> >>>>> http://www.ClueCon.com <http://www.cluecon.com/> >>>>> http://www.OSTAG.org <http://www.ostag.org/> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> 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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org <http://www.freeswitch.org/> >>> http://www.ClueCon.com <http://www.cluecon.com/> >>> http://www.OSTAG.org <http://www.ostag.org/> >>> >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >> >> _________________________________________________________________________ >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org <http://www.freeswitch.org/> > http://www.ClueCon.com <http://www.cluecon.com/> > http://www.OSTAG.org <http://www.ostag.org/> > > _________________________________________________________________________ > 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 > > >

1 0

FS as a centralized Voicemail system
by adahary 26 Jul '13

26 Jul '13

I'm running multiple FS servers with centralized DB for registration and routing tables. Any endpoint can dynamicaly register to any FS server. The FS servers should function as simple routers/switchers without handeling any media tasks like playing sound files or voicemail read/write messages. I would like to setup one FS server to function as a centralize Voicemail system for all users. I've search this mailing list and google arround with no help (all relate to local VM setup). Any advice/tip will be appriciated thanks Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-as-a-centralized-Voicemail… Sent from the freeswitch-users mailing list archive at Nabble.com.

4 10

What diagnostics are needed for a suspected memory leak?
by D D 26 Jul '13

26 Jul '13

Hi, I have been using sipp to send messages into FS for several days now. I am using the conferencing application, and playing some announcements into the conference. I have noticed a constant growth in memory of the FS process. I have verified that there are only 11 channels active (and in the conference), so I don't think it is an issue with the test scenario. What kind of diagnostics should I gather in order to submit this problem? Thanks, David

2 1

SIP dialogue mismatch when pinging FS with SIPp UA
by Victor Chukalovskiy 25 Jul '13

25 Jul '13

Hello, Freeswitch for some reason ignores UA's ACK and keeps re-transmitting 483 response (as per attached example). I suspect dialogue mismatch, but don't see anything abnormal in the ACK. From: To: and Call-ID: all match! Could someone with good eyes help find what makes FS unhappy about the ACK? Enabling debug output did not not reveal anything (FS does not print any debug output when receiving the ACK). SIP call flow: UA --INVITE--> FS UA <--100 Trying -- FS UA <--483-- FS UA --ACK--> FS And here is what I don't like: UA <--483-- FS UA <--483-- FS UA <--483-- FS .... UA is my SIPp that's set to ping FS with INVITE and expects 483 in response (or any rejection that I like). This is needed to monitor call processing (FS uses mod_lcr to return error channel like 483, so this will allow me to ping mod_sofia as well as mod_lcr). Any help is appreciated. Thank you, Victor

2 2

Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE
by smrdoshi 25 Jul '13

25 Jul '13

Hi Guys, I am trying to implement DID -> Local extension calling feature in FS. But all the time when i am calling DID number and its try to dial local extension number, we are getting NORMAL_TEMPORARY_FAILURE. I tried everything like changed codes, changed some settings in sip profile but nothing come up. It looks like some network topology can be the issue. Here is the log, http://pastebin.com/J8UhxQKt Can anybody please check and help me? Thanks in advance. Thanks, Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Originate-Failed-Cause-NORMAL… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Freeswitch not correctly terminating dialog with BYE for T.38 invite
by Bruce Lefko 25 Jul '13

25 Jul '13

Hi there! We have a situation, primarily with specific numbers, where freeswitch executes a T.38 INVITE and our SIP provider is responding with a 408 Request Timeout, and Freeswitch responds with an ACK. However, no BYE is sent and our SIP provider thinks the call is still active, and ends up billing us for a lot of dead air. Thankfully they have a timeout and stop billing us after 6 hours, but its still costing us a lot of money. The error Freeswitch is giving us in the cdr for this call is: RECOVERY_ON_TIMER_EXPIRE We've used the latest code from git as well as 1.2, and the issue persists. Furthermore, the SIP provider claims this is not correctly following the RFC:If the response for a request within a dialog is a 481 (Call/Transaction Does Not Exist) or a 408 (Request Timeout), the UAC SHOULD terminate the dialog. A UAC SHOULD also terminate a dialog if no response at all is received for the request (the client transaction would inform the TU about the timeout.) For INVITE initiated dialogs, terminating the dialog consists of sending a BYE. Thanks!

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