July 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_managed users?
by Michael Giagnocavo 13 Aug '09

13 Aug '09

Hey, if there are any mod_managed users on this list, I'd love it if you were able to let me know. I'd like to get feedback, positive or negative, on what worked, what didn't, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net Thanks! -Michael

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mod_php needed
by Christian Löschenkohl 09 Aug '09

09 Aug '09

hello i am working for an austrian voip carrier. for a few months i work with freeswitch and it is simply great. it solves our needs in many places (high volume, flexible, stable). the only thing i really miss is the avalibilty of php as a call control language. mod_php or mod_freehp are not compiling anymore and my c++ knowledge isn't that good (or even there :-) ). i know there is perl, i also implemented some applications (conference system with provisioning, inbound call routing to our application servers, some tests as pbx), but what should i say - perl and me aren't compatible in the end. the greatest thing for us would be that mod_php comes alive again with the functional state of mod_perl (http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes) there is also a bounty entry about mod_php, to pay for this would also be an option and could be discussed. keep on with the excellent work and greetings from austria -- Ing. Christian Löschenkohl Technische Leitung, Forschung& Entwicklung VoIP xpirio Telekommunikation& Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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Skypiax, Skype endpoint and trunk, revamped
by Giovanni Maruzzelli 08 Aug '09

08 Aug '09

Ciao FreeSWITCHers, please have a look at the much changed wiki page: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk and checkout and test the code in svn. Much has happened, various bug fixes and features added. Most notable: - multiple instances of the same Skype username on Linux (eg: running 20 concurrent channels as "Bob" Skype user) - adding and removing interfaces on the fly (patch sent by Muhammad Shahzad) - easier creation of Skype clients configuration directory - reduced latency - better robustness - running as Windows Service - customized ALSA driver for more devices with less IRQs and context switches - custom kernel tickless and 100HZ (eg. solves high load problems in CentOS and in virtualization) - interactive command line client for prototyping Also, please note that ALSA drivers version 1.0.20 seems to be much more stable in our kind of usage (snd-dummy). Various other enhancements will come, but in the mean time please give feedback on the current svn code (we want to be robust for the 1.0.4 Release :-) ) See you all at www.cluecon.com, talk on Skypiax August 4th at 4.30 pm ! -giovanni Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039

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ESL Wrapper
by Rex Alex 08 Aug '09

08 Aug '09

Hi All, I am new to freeswitch but installed(freeswitch version 1.0.3), configured and tested successfully. Now I want to do the dialling funtions through a php script. Read about Event Socket Library(ESL). How to implement the same in freeswitch. Please assist. Thanks, Rex

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Compact, fanless appliance?
by Fred-145 07 Aug '09

07 Aug '09

Hello For those SOHO customers (ie. at most, a couple of POTS/ISDN connections and simultaneous SIP calls) who'd rather not use a big, noisy PC to run Freeswitch, I'd like to offer an alternative that has the following features: - not old hardware sold on eBay, ie. it must be up-to-date hardware sold by a company currently in business - compact, silent - has room for a 2.5" hard-disk, but if not, must provide a CompactFlash plug - ideally, room for a PCI card, possibly laid down with a riser to save space - total cost (shipping + VAT) < 200 euro If it's cheaper and not much work, I don't mind buying the parts and putting the box together myself, but otherwise, I'd rather order a complete box, ready-to-use. What are my options to provide customers with that kind of solution? Thank you for any hint. -- View this message in context: http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23193738.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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pocketsphinx
by Helmut Kuper 05 Aug '09

05 Aug '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to change pocketsphinx's grammar from default (english) to german. I found this archive (http://www.repository.voxforge1.org/downloads/de/Trunk/AcousticModels/) which contains similar files like those which can be found in grammar/model/communicator directory. Unfortunately FS crashed without writing a core file nor logfile enries as soon as as pizza demo trys to detect speech. Any Ideas? Maybe someone has already working grammar/model files for german language? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKVzne4tZeNddg3dwRAthzAJ4hvonJLgaTWc3kCQXhESb2wsTu8QCeP/DD skUOkNgUHLRaKqVGWWk1uM8= =HkcN -----END PGP SIGNATURE-----

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Using tone_detect application
by Patrick Grondin 03 Aug '09

03 Aug '09

Hi, I'm doing some tests between 2 FS to understand how the tone_detect application works. I'm trying to detect a SIT tone, but I can't seem to detect the 3 tones. I only get the first activated tone. If I have all tones activated - - - > I detect only the first tone of my wav file. If I have 2nd segment tones and up activated - - - > I detect only the second tone of my wav file. If I have the 3rd segment tones activated - - - > I detect only the third tone of my wav file. I see that tone_detect can detect all 3 tones, but never at the same time. Does anyone have an idea of what I could be doing wrong ? Thanks ! I'm using FreeSWITCH Version 1.0.trunk (14397). My dialplan looks like this : <extension name="incoming"> <condition field="destination_number" expression="^(514[0-9][0-9][0-9][0-9][0-9][0-9][0-9])$"> <action application="answer"/> <action application="tone_detect" data="SIT1LO 913.8 w +25000 set SIT1LO=true"/>  <action application="tone_detect" data="SIT1HI 985.2 w +25000 set SIT1HI=true"/>  <action application="tone_detect" data="SIT2LO 1370.6 w +25000 set SIT2LO=true"/>  <action application="tone_detect" data="SIT2HI 1428.5 w +25000 set SIT2HI=true"/>  <action application="tone_detect" data="SIT3LO 1776.7 w +25000 set SIT3LO=true"/>  <action application="tone_detect" data="SITITU1 950 w +25000 set SITITU1=true"/>  <action application="tone_detect" data="SITITU2 1400 w +25000 set SITITU2=true"/>  <action application="tone_detect" data="SITITU3 1800 w +25000 set SITITU3=true"/>  <action application="bridge" data="sofia/internal/$1(a)192.168.100.9"/> </condition> </extension>

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FS beats Aculab Prosody S on subjective test on lay users for conference quality
by Fernando Testa 03 Aug '09

03 Aug '09

Hello freeswitchers! I thought you're be pleased to know: FS beated Prosody S on subjective test on lay users. After 3 users testing the two conferences and reporting far better quality on FS, we decided to make a more rigorous test. Then we made a blind test by asking 4 lay users to dial two urls for 10 minutes from their x-lite and report the results. These users are non-techie and never heard about FS at all. The first url was Aculab Prosody S test app, then, a second call to FS on default conf (extension 3000). All users preferred FS. Some users were almost enthusiastic after entering FS conf. Some even started to blame ProsodyS after hearing FS. The average report told 50% more satisfaction on FS. Questions on report were simple ones, with results from 1(worse) to 5(best) : 1-Overall opinion(1-crap; 5-awesome); 2-Delay (1-more delay, 5-no delay at all); 3-Audio quality (1-noisy, choppy, etc; 5-cristal clear); Machines used: Prosody S: Windows Vista Business, Intel Core 2 Duo E7400 @ 2.8 GHz, 4Gb de RAM FreeSwitch: CentOS 5.4, Intel(R) Pentium(R) 4 CPU 2.40GHz, 1Gb RAM Congratulations! You made a great product! -- Fernando Gregianin Testa Voice Technology Ltda +55 11 35882166

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Language of the speech
by Rudolf Denert 03 Aug '09

03 Aug '09

Hi again, does anybody know why my freeswitch doesn't play German speech-files? Here is my construct: I'm generating a random number in my lua script: rand = math.random(11, 1000); The I want that the freeswitch says the "random number": session:execute("say", "de name_spelled iterated " ..test_nummer); It works fine in English but I don't bring the freeswitch to say the number in german. I installed the freeswitch-lang-de_1.0.3-1_i386.deb, change the line in freeswitch.xml: <section name="phrases" description="Speech Phrase Management"> <macros> <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/> </macros> </section> Here is an extraction from the fs_cli: 2009-07-31 16:23:47 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/external/anonymous(a)anonymous.invalid Execute set(default_language=de) 2009-07-31 16:23:47 [DEBUG] mod_dptools.c:711 set_function() sofia/external/anonymous(a)anonymous.invalid SET [default_language]=[de] But I don't here any German speech only the English on Does anybody of you have an idea? Thanks a lot again. Best regards -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert(a)tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enthält vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet.

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LUA: Independent control of each call leg.
by peely 01 Aug '09

01 Aug '09

Hi, I'm trying to develop an application using lua and need to control the inbound and outbound legs independently, even when they are switched together. I can initiate the outbound session but I can't seem to bridge without losing control of the script. For example, if I use: local api = freeswitch.API(); inSession = session; inSession:answer(); inSession:setAutoHangup(false); egSession = freeswitch.Session("sofia/default/mynum(a)mydomain.com"); egSession:setAutoHangup(false); if egSession:ready() then api:execute("uuid_bridge",inSession.uuid .. " " .. egSession.uuid); end while egSession:ready() do inSession:sleep(1000); end Then I lose the script entirely, and if I use: inSession:execute("bridge", "sofia/default/mynum(a)mydomain.com") Then I lose the ability to control the call whilst the outbound is in progress. Does anyone know a way I can allow ingress to egress calling whilst still maintaining script control mid-call? I also need to ingress to hear provisional speech during outbound connect. I've looked at conferencing but there seems to be quite a lot of automated messaging. Thanks, Neil. -- View this message in context: http://www.nabble.com/LUA%3A-Independent-control-of-each-call-leg.-tp247440… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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