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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 > > >

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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.

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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

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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

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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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error in compiling freeswitch 1.2.7
by Ashish Mishra 25 Jul '13

25 Jul '13

hello, i have downloaded freeswitch installed the prerequisites: gnumake automake autoconf libtool gcc wget libncurses bzip2 and also have successfully done the following steps: run the bootstrap.sh edit modules.conf run the configure script BUT WHEN TYPING THE COMMAND make && make install i am getting the following error ./.libs/libfreeswitch.so: undefined reference to `inflateReset' ./.libs/libfreeswitch.so: undefined reference to `inflateEnd' ./.libs/libfreeswitch.so: undefined reference to `inflateInit2_' ./.libs/libfreeswitch.so: undefined reference to `inflate' ./.libs/libfreeswitch.so: undefined reference to `zlibVersion' ./.libs/libfreeswitch.so: undefined reference to `inflateInit_' collect2: ld returned 1 exit status make[1]: *** [freeswitch] Error 1 make: *** [all] Error 2 kindly help me Ashish

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Remote media gateway
by Ashwin Rath 25 Jul '13

25 Jul '13

Is there a way to configure FreeSWITCH such that it places an external media gateway in the media path when it acts like an SBC ? -- Ashwin Kumar Rath

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Restarting sofia
by Alex Lake 25 Jul '13

25 Jul '13

Is there a way to quickly restart all sofia gateways such that acl.conf will be re-read?

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Freeswitch not responding to sip messages from call out leg
by Zvi Agmon 25 Jul '13

25 Jul '13

Hello, We are using FS to bridge incoming call to an outbound telephone through a gateway provider. The bridge is created OK, but we found that after 3 minutes FS is no longer responding to sip messages from the call out provider. Thus, in case the provider sends sip-Invite message, it gets no response from FS and disconnect the call out leg. Note that FS is also not aware of the Bye message so leg A is still on the line and when it hangs up FS send Bye message to leg B and get as response "Unknown Dialog". I set the siptrace on and see that no packets arriving sofia (when the calling leg hanged up I do see the packets). Also I set all sofia components log level to 7 - this is what was printed out after the call answered: 2013-07-08 10:28:49.661014 [NOTICE] sofia.c:6288 Channel [sofia/external/97249126793] has been answered 2013-07-08 10:28:49.661014 [DEBUG] switch_channel.c:3291 sofia/external/97249126793 process sched_hangup +7199 DJIWR4KKCZB3DGAS7LYLI76IPE(a)81.201.82.45 alloted_timeout: sched_hangup(+7199 DJIWR4KKCZB3DGAS7LYLI76IPE(a)81.201.82.45 alloted_timeout) 2013-07-08 10:28:49.661014 [DEBUG] switch_scheduler.c:214 Added task 6 switch_ivr_schedule_hangup (DJIWR4KKCZB3DGAS7LYLI76IPE(a)81.201.82.45) to run at 1373275728 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/external/97249126793 [BREAK] 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/internal/972522977131(a)voxbone.com [BREAK] nta.c:8950 outgoing_timer_dk() nta: timer D fired, terminate INVITE (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/1 term, 1/2 free nta.c:8831 outgoing_timer_bf() nta: timer F fired, terminating ACK (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout, 0/0 term, 1/1 free We are running FreeSWITCH Version 1.2.7+git~20130307T054051Z~0a2e713593 (git 0a2e713 2013-03-07 05:40:51Z) On ubuntu release 12.04 Please help. Thanks Zvi Agmon Best regards Zvi Agmon Lexifone email: zvi(a)lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109

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