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metadigits
by Sam 13 Jan '11

13 Jan '11

Any Ideas, Why I am getting this logs below:- 2011-01-13 15:13:51.821614 [DEBUG] switch_rtp.c:3105 RTP RECV DTMF #:960 2011-01-13 15:13:53.001254 [DEBUG] switch_rtp.c:3105 RTP RECV DTMF 3:1280 2011-01-13 15:13:53.001254 [WARNING] switch_ivr_async.c:2882 sofia/internal/ 7001(a)192.168.2.190 Ignoring meta digit '3' not mapped I have in my dialplan :- <action application="set" data="bind_meta_key=#"/> <action application="bind_meta_app" data="1 ab s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 ab s execute_extension::att_xfer XML features"/> <action application="bind_meta_app" data="3 ab s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> the other (1&2) meta keys are working. Regds Sam

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Freeswitch eLearning
by Deon Vermeulen 13 Jan '11

13 Jan '11

Hi Does anyone know of an online training course for Freeswitch? Thank you very much Kind Regards Deon

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Compile error
by Sam 13 Jan '11

13 Jan '11

Hello , Got the compile error for latest git , making all mod_spandsp Making all in src libtool: compile: gcc -DHAVE_CONFIG_H -I. -I.. -I/usr/include/libxml2 -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -msse2 -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -MT v17rx.lo -MD -MP -MF .deps/v17rx.Tpo -c v17rx.c -fPIC -DPIC -o v17rx.o v17rx.c: In function âv17_rx_symbol_timing_correctionâ: v17rx.c:169: error: âRX_PULSESHAPER_COEFF_SETSâ undeclared (first use in this function) v17rx.c:169: error: (Each undeclared identifier is reported only once v17rx.c:169: error: for each function it appears in.) v17rx.c: In function âequalizer_restoreâ: v17rx.c:229: error: âRX_PULSESHAPER_COEFF_SETSâ undeclared (first use in this function) v17rx.c: In function âequalizer_resetâ: v17rx.c:250: error: âRX_PULSESHAPER_COEFF_SETSâ undeclared (first use in this function) v17rx.c: In function âv17_rxâ: v17rx.c:1151: error: âRX_PULSESHAPER_COEFF_SETSâ undeclared (first use in this function) v17rx.c:1162: error: ârx_pulseshaper_reâ undeclared (first use in this function) v17rx.c:1181: error: âRX_PULSESHAPER_GAINâ undeclared (first use in this function) v17rx.c:1198: error: ârx_pulseshaper_imâ undeclared (first use in this function) v17rx.c: In function âv17_rx_fillinâ: v17rx.c:1235: error: âRX_PULSESHAPER_COEFF_SETSâ undeclared (first use in this function) v17rx.c: In function âv17_rx_restartâ: v17rx.c:1368: error: âRX_PULSESHAPER_GAINâ undeclared (first use in this function) make[7]: *** [v17rx.lo] Error 1 make[6]: *** [all] Error 2 make[5]: *** [all-recursive] Error 1 make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la] Error 2 make[3]: *** [mod_spandsp-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Regds Sam

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Re: [Freeswitch-users] deflect question
by Madovsky 13 Jan '11

13 Jan '11

I try with different SIP phones and apparently it works with x-lite v4. Is REFER message usually supported as standard ? ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Wednesday, January 12, 2011 10:34 PM Subject: Re: [Freeswitch-users] deflect question I'm sure also that the call is answered before deflect ----- Original Message ----- From: Brian West To: FreeSWITCH Users Help Sent: Wednesday, January 12, 2011 8:40 PM Subject: Re: [Freeswitch-users] deflect question try making it sip:something@blah /b On Jan 12, 2011, at 4:50 PM, Madovsky wrote: I'm stucked. in the log the call stalls with this line EXECUTE sofia/internal/5555555555555(a)myfs.org deflect(9999999999999(a)otherfs.org) big silenece and nothing happens the otherFS doesn't receive anything. I tried to call manually the deflect sip and it works. Did I forget something ? thanks ---------------------------------------------------------------------------- _______________________________________________ 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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Re: [Freeswitch-users] deflect question
by Madovsky 13 Jan '11

13 Jan '11

I'm stucked. in the log the call stalls with this line EXECUTE sofia/internal/5555555555555(a)myfs.org deflect(9999999999999(a)otherfs.org) big silenece and nothing happens the otherFS doesn't receive anything. I tried to call manually the deflect sip and it works. Did I forget something ? thanks ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Wednesday, January 12, 2011 4:36 PM Subject: Re: [Freeswitch-users] deflect question I mean is the request arrives as a new INVITE ? ----- Original Message ----- From: Brian West To: FreeSWITCH Users Help Sent: Wednesday, January 12, 2011 4:22 PM Subject: Re: [Freeswitch-users] deflect question What exactly is the question? /b On Jan 12, 2011, at 3:08 PM, Madovsky wrote: if I use <action application="deflect" data="sip:someone@otherFS.com"/> in the otherFS dialplan should I createan extension with "someone" condition only ? Thanks ---------------------------------------------------------------------------- _______________________________________________ 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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FreeSwitch softphone testing
by Jack Loranger 12 Jan '11

12 Jan '11

I listened in on the conference this morning and if Mitch would like someone to use his phone I would be happy to use it while I am testing my FreeSwitch app. Can it be used in a webpage? Jack

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mod_conference end
by Madovsky 12 Jan '11

12 Jan '11

How can a xml dialplan guess that a conference is ended ? I mean the last participant left ? Thanks

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ZORG, new C++ and Java ZRTP implementation public release
by Fabio Pietrosanti (naif) 12 Jan '11

12 Jan '11

Milan, January 12th 2011 - PrivateWave Italia S.p.A, italian company engaged in developing technologies for privacy protection and information security in voice telecommunications, is pleased to announce the release of ZORG, a new open source ZRTP protocol implementation available for download from http://www.zrtp.org . ZRTP [1] provides end-to-end key exchange with Elliptic Curve Diffie-Hellmann 384bit and AES-256 SRTP encryption . ZORG has been originally developed and implemented in PrivateWave's PrivateGSM voice encryption products available for the following platforms: Blackberry, Nokia and iOS (iPhone) . Zorg C++ has been integrated with PJSIP open source VoIP SDK [2] and it's provided as integration patch against PJSIP 1.8.5. It has been tested on iPhone, Symbian, Windows, Linux and Mac OS X. Zorg Java has been integrated within a custom version of MJSIP [3] open source SDK on Blackberry platform and it includes memory usage optimizations required to reduce at minimum garbage collector activity. Both platforms have separated and modular cryptographic back-ends so that the cryptographic algorithms implementation could be easily swapped with other ones. ZORG is licensed under GNU AGPL and source code is available on github athttps://github.com/privatewave/ZORG. We are releasing it under open source and in coherence with our approach to security [4] as we really hope that it can be useful for the open source ecosystem to create new voice encryption systems in support of freedom of speech. More than 20 pjsip-based open source VoIP encryption software (several written in Java) could directly benefit from ZORG release. We would be happy to receive proposal of cooperation, new integration, new cryptographic back-ends, bug scouting and whatever useful to improve and let ZRTP affirm as voice encryption standard. Zorg is available from http://www.zrtp.org . [1] ZRTP: http://http://en.wikipedia.org/wiki/ZRTP [2] PJSIP: http://www.pjsip.org [3] MJSIP: http://www.mjsip.org [4] Security approach:http://www.privatewave.com/security/approch.html

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Vestec Speech Engine: ASR 2.1 Release
by Kashif Kahn 12 Jan '11

12 Jan '11

Dear All, We have launched a major upgrade to our ASR engine that offers the best deal around for enabling speech recognition with "command and control" type IVR applications. The new architecture boasts a number of advancements over version 1.1, including: - Improved US English acoustic model - DTMF recognition - MRCP support v1 and v2 - Highly scalable architecture and Redundancy - C++ API availability - SRGS-XML (.grxml) grammar support - Improved logging - Bug Fixes The engine can be integrated with different contact center and soft-PBX platforms (such as Freeswitch) using MRCP interface. A starter kit comprising a specially priced full-function engine is available for $25 while a regular one channel (ie. port) license can be purchased for $99. Please visit Vestec webstore: http://www.vestec.com/ Regards, -Kashif -- Kashif Kahn VP Business Development Vestec Inc Waterloo, ON Canada phone: +1 519 885-7615

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FreeSWITCH Conference Call Today
by Michael Collins 12 Jan '11

12 Jan '11

Hello all, We have a light agenda today: http://wiki.freeswitch.org/wiki/FS_weekly_2011_01_12 Bring your questions and ideas and be ready to share! -MC

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