January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Retrieving hangupCause from b-leg in mod_lua
by Benoit Raymond 23 Jan '13

23 Jan '13

Hello, Is there a way in lua to figure out why the b-leg hung up? I have an inbound call that I need to bridge so when lua is invoked, there is a session created for the inbound call and then I use session:execute("bridge", "the b-leg path") and either the call is bridged or call_timeout kicks in. I need to continue my script based on the hangup cause whether the b-leg produced a NO_ANSWER, USER_BUSY or any other condition. The session:hangupCause is of no use since my session is still active. I found the variable "bridge_hangup_cause" on the wiki but it returns nil if I do session:getVariable("bridge_hangup_cause"). I am using 1.3.13 Thanks much. Benoit Raymond AMI Concept Inc. Business VoIP Solutions / Web Design Phone: (450) 553-1231 http://www.amiconcept.com <http://www.amiconcept.com/>

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DTMF bug report
by Richard Gration 23 Jan '13

23 Jan '13

Hi, I submitted a bug report today (FS-5047). I think I should have raised it at a slightly higher priority, it's more than minor. Is there anything I can do about that? Cheers, Rich -- Once our basic material needs are met - in my utopia, anyway - life becomes a perpetual celebration in which everyone has a talent to contribute. But we cannot levitate ourselves into that blessed condition by wishing it. We need to brace ourselves for a struggle against terrifying obstacles, both of our own making and imposed by the natural world. And the first step is to recover from the delusion that is positive thinking. -- Barbara Ehrenreich

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Re: [Freeswitch-users] mod_stress
by Raimund Sacherer 23 Jan '13

23 Jan '13

Hi Peter, can you also make some statistics available on how much this module, if activated, adds to the ressource usage? If you have a busy callcenter, would this make a noticable impact on CPU usage? Also, would this use an additional G729 codec for calls using this codec? thanks, best regards Raimund ----- Original Message ----- From: "Peter Steinbach" <lists(a)telefaks.de> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Miércoles, 23 de Enero 2013 0:34:41 Subject: Re: [Freeswitch-users] mod_stress Thanks Steven, I got it working and reported my howto and findings in the wiki: http://wiki.freeswitch.org/wiki/Mod_stress I found it to be a nice application, but as you explained, there are some false positives. Best regards Peter On 01/22/13 12:39, Steven Ayre wrote: Media bugs are used to hook into the call and provide decoded audio to the module. That requires transcoding, naturally. See stress_callback which is called for each audio frame plus some init/cleanup hooks, and read up on the API at http://docs.freeswitch.org/group__mb1.html . I suspect read vs write would correspond to the speaking/listening directions of the channel. Read from caller, write to caller? The names come from the media bug API. The read/write options will add the corresponding media bug, stop will remove them. SWITCH_ADD_APP tells you it's a dialplan application - it's started from dialplan, although you can also trigger dialplan apps on a channel via ESL using uuid_broadcast. The FFT algorithm is used to extract the frequencies present in the sample (the power spectrum). The algorithm then tries to detect stressed based on that. Looks like the algorithm at least in part attempts to look for tremor in the voice. More tremor indicates more stress. Over a threshold it raises an event. Bear in mind that as it's pattern recognition (and little tested at that) you'll get a lot of false positives and false negatives. It's also looking for something in the voice that may or may not be there - some people might be able to speak with an unstressed voice when they're actually quite stressed. -Steve On 22 January 2013 11:19, Peter Steinbach < lists(a)telefaks.de > wrote: <blockquote> Hello Anthony, although it may be incomplete, it may be useful for us. Can you give us some basic explanation, how this works? I can see in the code that there is a stress command with read|write|stop parameter. Is this to be issued via event socket or is it a dialplan application? I suspect that "read" somehow starts the detection and "stop" somehow stops the detection. For "write" I currently have no idea. I would be happy to test this and report my configuration, explanations and examples in the wiki. Best regards Peter On 01/22/13 00:30, Anthony Minessale wrote: <blockquote> That module is incomplete we made it several years ago to test some code that was supposed to pick up on stress in your voice in a media bug so we could fire events and whisper "lie" to the caller as he talked to the guy. On Mon, Jan 21, 2013 at 4:56 PM, Peter Steinbach < lists(a)telefaks.de > wrote: <blockquote> I am also wondering, what this module does in detail. Regarding the source code, I can see that FFT's are made in order to to get some sort of frequency profile. But the code and remarks do not give me a clue how this module actually works. Best regards Peter On 05/26/12 15:05, BF wrote: > Hello, > > I'd like to know what the "stress write" from mod_stress module does. I have hard time finding documentation or an example about mod_stress module. > > Anyone? > > Thank you. > > Eric > _________________________________________________________________________ > 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 > > Join Us At ClueCon - Aug 7-9, 2012 > > 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 > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto: lists (att) telefaks.de Internet: www.telefaks.de _________________________________________________________________________ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn: +19193869900 _________________________________________________________________________ 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 </blockquote> -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de _________________________________________________________________________ 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 </blockquote> _________________________________________________________________________ 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 </blockquote> -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de _________________________________________________________________________ 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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FreeSWITCH Community Conference Call Today
by Michael Collins 23 Jan '13

23 Jan '13

Hello folks, On today's call we have a few news items to discuss, after which we will have an open forum for Q&A on FreeSWITCH topics. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_23 Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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How to test Freeswitch before moving to production?
by Yehavi Bourvine 23 Jan '13

23 Jan '13

Hello, We are running version 1.0.6; we did not upgrade for a long time due to some problems (mainly stuck BLFs) with the newer versions which have been corrected a few months ago. A few weeks ago I pulled the latest GIT and did functional testings on our development system; all was OK. Yesterday I moved our production system to that version. After a few hours the problems started: stuck BLFs, hangups, LUA scripts sometimes do not run. I had to revert back to the old version. My question: Is there some simple test suite which I can test various scnarios on my development system? like, "call XX, while this call is active call YY which is monitored by XX and hangup", etc. It is hard to cover all cases manually, so I would like something automated. Thanks, __Yehavi:

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FreeSwitch Multitenant registrar and generic SIP PBX
by Andrea Mucci 23 Jan '13

23 Jan '13

Hi All I have a project for one university, this is a free project and i want to help him to solve some thinks. the idea is create a sytem multitenant or multicompany for this university. the IP PBX work with sip and don't have any multitenant service. My idea is use the Multitenat service of Fresswitch and all other services like RTP media etc with the IP PBX. Is this possible? The schema will be., UA send a register "(1000(a)test.com) to Freeswitch Freeswitch register with the SIP PBX with the extension 2000. calls and services are mediated by the SIP PBX. UA send a register "1001(a)test.co to freeswith Freeswitch send a register request to the SIP PBX to the 2001 extension calls and services are mediated by the SIP pBX sorry if you tink this is a supid question but i'm newbie and i want to help this university with this project. have a nice day

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Re: [Freeswitch-users] mod_gsmopen with Motorola i465
by DanB 23 Jan '13

23 Jan '13

Giovanni, Thank you for your so fast answer. Do I have any chance finding a CDMA phone working with your module then? DanB

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mod_gsmopen with Motorola i465
by DanB 23 Jan '13

23 Jan '13

Hey Guys (especially Giovanni), I am trying to use mod_gsmopen - the mobigater flavor - with a Motorola i465. Before opening a jira I was wondering if anyone had any experience with this phone and if there are chances to make it work? I can see some information being exchanged when I try to dial out via originate command but it fails at some point after sending "AT+MMGL="HEADER ONLY" (see bellow). I should mention that the phone is CDMA version. Further logs/debug available on request. """ 2013-01-23 11:19:50.193765 [DEBUG] gsmopen_protocol.c:2360 rev [(nil)|37 ][DEBUG_GSMOPEN 2360 ][interface1][-1,24, 3] sending: AT+MMGL="HEADER ONLY" 2013-01-23 11:19:50.233538 [DEBUG] gsmopen_protocol.c:1078 rev [(nil)|37 ][DEBUG_GSMOPEN 1078 ][interface1][-1,24, 3] Read line 0: |ERROR| 2013-01-23 11:19:50.233538 [DEBUG] gsmopen_protocol.c:1943 rev [(nil)|37 ][DEBUG_GSMOPEN 1943 ][interface1][-1,24, 3] got ERROR 2013-01-23 11:19:50.233538 [WARNING] gsmopen_protocol.c:3911 rev [(nil)|37 ][WARNINGA 3911 ][interface1][-1,24, 3] AT+MMGL="HEADER ONLY" does not get OK from the phone. If your phone is not Motorola, please contact the gsmopen developers. Else, if your phone IS a Motorola, probably a long msg was incoming and ther first part was read and then deleted. The second part is now orphan. If you got this warning repeatedly, and you cannot correctly receive SMSs from this interface, please manually clean all messages (and the residual parts of them) from the cellphone/SIM. Continuing. """ Thanks in advance for any kind of tip! DanB

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Make a call with Perl
by Matthias Cramer 23 Jan '13

23 Jan '13

Hi All I try to make a Perl Script which calls a certain external Phone number. It then should wait until someone picks up and play a sound file The user on the other end then needs to confirm the call with a pin Then another sound file should be played and the the call should be terminated. I tried several ways with ESL and originate. But I allays end up with a call I have no control over. I can read DTMF but can not hang up the call or play a second file. I think I am going in to a wrong direction. Any hints ? Regards Matthias -- Matthias Cramer / mc322-ripe Senior Network & Security Engineer iway AG Phone +41 43 500 1111 Badenerstrasse 569 Fax +41 44 271 3535 CH-8048 Zürich http://www.iway.ch/ GnuPG 1024D/2D208250 = DBC6 65B6 7083 1029 781E 3959 B62F DF1C 2D20 8250

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Handling call forward activate / deactivate through FreeTDM using Sangoma A200
by Benoit Raymond 23 Jan '13

23 Jan '13

Hello FS users, I have Freeswitch (1.3.13) using a Sangoma A200 card with 2 FXO. I am using FreeTDM and inbound/outbound calls are going through. One of the POTS line has call forwarding feature so my customer wants to have the ability to dial *72 / *73 and get proper dialtone back from the phone company so he can activate or deactivate CFW. While the call is marked as answer in the log, I dont hear the dialtone back so unsure if it works or not. This is the dialplan I used: <extension name="CFW Handler"> <condition field="destination_number" expression="^(\*72|\*73)$"> <action application="set" data="default_language=${default_language}"/> <action application="bridge" data="freetdm/FXO/1/$1"/> <action application="hangup"/> </condition> </extension> Is there a better way or a good way of doing this? Thanks, Benoit Raymond AMI Concept Inc. Solutions VoIP Affaire / Création Web Tel: (450) 553-1231 http://www.amiconcept.com <http://www.amiconcept.com/>

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