June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

eavesdrop extension condition in default.xml?
by Lars Zeb 17 Jun '09

17 Jun '09

In conf/dialplan/default.xml, the eavesdrop extension's condition is - expression="^88(.*)$|^\*0(.*)$"> Is this intended? I thought it was defined to eavesdrop on internal extensions. Why wouldn't it be something like - expression="^88(\d{3,5})$|^\*0(\d{3,5})$">? The way it is now, if I dial 888-555-1212 (rather than 1-888-555-1212), it goes into the eavesdrop extension. Log: 1030 Dialplan: sofia/internal/1009(a)192.168.10.29 parsing [default->eavesdrop] continue=false 1031 Dialplan: sofia/internal/1009(a)192.168.10.29 Regex (PASS) [eavesdrop] destination_number(8885819795) =~ /^88(.*)$|^\*0(.*)$/ break=on-false 1032 Dialplan: sofia/internal/1009(a)192.168.10.29 Action answer() Thanks, Lars

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Fwd: [UniMRCP] Open source ASR and TTS plugins
by Arsen Chaloyan 17 Jun '09

17 Jun '09

Mitul, >2) What are the deliverable we expecting of Arsen's effort? And a generic ETA of those deliverables? Both PocketSphinx and Flite will be integrated into open source UniMRCP server as ASR and TTS plugins respectively and will be publicly available under the Apache 2.0 license as UniMRCP itself. It's matter of 2~3 weeks to have basically working and usable plugins for them. Thank you, Arsen. From: Mitul Limbani <mitul(a)enterux.com> Date: June 17, 2009 1:50:52 PM CDT To: "freeswitch-dev(a)lists.freeswitch.org" <freeswitch-dev(a)lists.freeswitch.org> Cc: "freeswitch-dev(a)lists.freeswitch.org" <freeswitch-dev(a)lists.freeswitch.org>, "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] [Freeswitch-dev] Fwd: [UniMRCP] Open source ASR and TTS plugins Reply-To: freeswitch-users(a)lists.freeswitch.org Brain, I can chip in $150 of my own as well, but two things: 1) what happens to mod_pocketsphinx ?? (is scribbleJ going to work on it?) 2) What are the deliverable we expecting of Arsen's effort? And a generic ETA of those deliverables? Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 17-Jun-09, at 23:35, Brian West <brian(a)freeswitch.org> wrote: Guys I'm tossing in $250 dollars of my own money on this ... who is going to pitch in? Arsen and I have been talking about how to accomplish this so we'll have an Open Source Speech Server via MRCP. Who wants to pitch in please paypal brian(a)freeswitch.org and I can wire it to Arsen. Thanks, Brian Begin forwarded message: ----- Forwarded Message ---- From: Arsen Chaloyan <achaloyan(a)gmail.com> To: UniMRCP <unimrcp(a)googlegroups.com> Sent: Wednesday, June 17, 2009 10:57:30 PM Subject: [UniMRCP] Open source ASR and TTS plugins Anybody interested in the development of open source ASR and TTS plugins for UniMRCP server write me offlist. PocketSphinx (ASR) - http://www.speech.cs.cmu.edu/pocketsphinx/ Flite (TTS) - http://www.speech.cs.cmu.edu/flite/ Thanks, -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org _______________________________________________ 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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Solaris 10 - Install Instructions Needed
by Kevin Green 17 Jun '09

17 Jun '09

I was wondering if anyone had a set of detailed step-by-step instructions on how to get FS up and running on Solaris 10 from a base Solaris install. I understand that there are a few pieces required to be able to build FS as described on the wiki ( http://wiki.freeswitch.org/wiki/Installation_Guide#Solaris) but it skips over the details of each step. If someone could fill in the details or provide me with some help it would be greatly appreciated. Also, if there are any differences between working on an x86 vs a SPARC platform it would be good to have notes on the different steps that should be taken including any configuration variables that would optomize FS for x86 or SPARC CoolThreads machine. I have attached a whitepaper describing the benefit of UltraSPARC T2 processors running VoIP applications for anyone's interest. This whitepaper is my motivation for wanting to try FS on Solaris. Regards, Kevin Green JohnnyVoIP 350 Legget Drive Kanata, ON, Canada K2K 2W7 Phone: 613 271 5993 Ext 1203

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javascript session.execute
by Baskar 17 Jun '09

17 Jun '09

*Hi, I have some questions can any one assist me and answer this query Question 1: Can we able to execute all the api command through JavaScript using session.execute. Question 2: How to kill the session using uuid_kill whether it is possible. **If yes means how we will use uuid_kill in session* * i have tried some thing like this : session.execute("uuid_kill", "session.uuid"); session.execute("uuid_kill session.uuid"); but there is no output for the session.execute. Whether it is possible to execute uuid_kill in javascript Thanks in advance Warm Regards, N.Baskar *

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session.getDigits() not working
by paul.degt＠gmail.com 17 Jun '09

17 Jun '09

Trying out latest trunk ans seems like js function session.getDigits() stopped working (not collecting any digits), I do see switch_rtp.c:1560 Send end packet for [5] ts=2222260 dur=2080/2080/2000 seq=8732 in debug log so I assume dtmf is ok. Anybody can shed some light on why wouldn't it work now? Works just fine under 1.0.3 release. I use slightly modified version of disa.js from fs examples. Thanks.

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Re: [Freeswitch-users] Compiling Issues: Opal with Latest SVN Builds 6-19-09 (Jon DiVita)
by Jonathan DiVita 17 Jun '09

17 Jun '09

I was able to compile mod_opal with the patches you suggested. Now, once that is done is there anything more I need to do inorder for mod_opal to show up when I run freeswitch? I didn't see that it had started when I ran freeswitch after the mod_opal compile. Thanks! Jon ----- Original Message ----- From: <freeswitch-users-request(a)lists.freeswitch.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Wednesday, June 17, 2009 12:26 AM Subject: Freeswitch-users Digest, Vol 36, Issue 159 > Send Freeswitch-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: Compiling Issues: Opal with Latest SVN Builds 6-19-09 > (Brian West) > 2. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 3. Re: session.getDigits() not working (Brian West) > 4. Re: How can I join two freeswitch on two servers? (Brian West) > 5. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 6. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > 7. Re: How can I join two freeswitch on two servers? (Brian West) > 8. Re: How can I join two freeswitch on two servers? (Edmar Cruz) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 16 Jun 2009 22:12:01 -0500 > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Compiling Issues: Opal with Latest SVN > Builds 6-19-09 > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <A8AB9B1C-35A3-4D25-82E7-D9F6E73EFBEC(a)freeswitch.org> > Content-Type: text/plain; charset="iso-8859-1" > > please see MODOPAL-10 on jira. > > /b > > On Jun 16, 2009, at 10:05 PM, Jonathan DiVita wrote: > >> Hello, all. I'm currently playing around with a new install of >> Freeswitch and wanted to try out mod_opal. Below are the current >> SVN builds for opal, ptlib, and freeswitch. I end up with the >> following errors when compiling. >> >> making all mod_opal >> Compiling mod_opal.cpp... >> Compiling mod_opal.cpp ... >> In file included from mod_opal.cpp:25: >> mod_opal.h:151: error: conflicting return type specified for >> ?virtual OpalLocalConnection* >> FSEndPoint::CreateConnection(OpalCall&, void*)? >> /usr/include/opal/opal/localep.h:267: error: overriding ?virtual >> ptlib_virtual_function_changed_or_removed****** >> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)? >> mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&, >> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*, >> switch_channel_t*)?: >> mod_opal.cpp:564: error: no matching function for call to >> ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, >> NULL)? >> /usr/include/opal/opal/localep.h:290: note: candidates are: >> OpalLocalConnection::OpalLocalConnection(OpalCall&, >> OpalLocalEndPoint&, void*, unsigned int, >> OpalConnection::StringOptions*, char) >> /usr/include/opal/opal/localep.h:276: note: >> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) >> make[4]: *** [mod_opal.lo] Error 1 >> make[3]: *** [all] Error 1 >> make[2]: *** [mod_opal-all] Error 1 >> make[1]: *** [mod_opal] Error 2 >> make: *** [mod_opal] Error 2 > >

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Porta Billing?
by Mark Campbell-Smith 17 Jun '09

17 Jun '09

Hi! Does freeswitch support extracting the billing data (PortaBilling) in SIP messages? If so, is there anyway I can get that information to an extension? 03:36:00.245: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDis playMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP sip.mydomain.com:5060;branch=z9hG4 bK1FF90 From: {sip:61XXXXXXXXX@sip.pennytel.com}; tag=47E011-580 To: {sip:61XXXXXXXXX@sip.pennytel.com}; tag=adfde4bc91cd85e752cb0672816ac1 a6-eb1b Call-ID: EE04FAB6-F24011DC-803AA58E-D5912FB 7 CSeq: 3 REGISTER PortaBilling: available-funds:7.37 currency:AUD Contact: {sip:61XXXXXXXXX@sip.mydomain.com:5060}; expires=3595 Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 Thanks!

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Freeswitch / Webserver
by Rudolf Denert 17 Jun '09

17 Jun '09

Hello, I have one question about including a webserver. I made a dialplan which I deposited on a webserver. That was no the problem. But I have a commandline in the dialplan that doesn't work correct. The fs_cli says: <action application="read" data="1 11 $${base_dir}/sounds/de/conf-getconfno.gsm raum 5000 #"/> (this line has no problem. I just mentioned this for the sake of completeness) <action application="execute_extension" data="test XML features"/> My features.xml looks like this: <extension name="test"> <condition field="destination_number" expression="^.*$"/> <condition field="${raum}" expression="^18404$" break="on-true"> <action application="phrase" data="spell,${raum}"/> <action application="conference" data="hans@test+112233"/> </condition> In my freeswitch server this line works perfekt. The extension "test" in features.xml is executed. For near information: When I call the FS (it is a conferencing system) the machine looks in the dialplan which is deposited on a webserver. Than I press the digits into phone to get access in the conferencing room. But than I get two important messages in the fs_cli: 2009-06-17 15:38:21 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 123456789->test in context features 2009-06-17 15:38:21 [WARNING] mod_dialplan_xml.c:263 dialplan_hunt() Context features not found Does anybody has an idea? Greetz -- -- 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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Force SIP UA to pick up call during ringing?
by Peter P GMX 17 Jun '09

17 Jun '09

I have managed to have a realtme status of a phone on a web page with event_socket and a push service to the web bowser. What I am now trying to do is roughly the following: * when a call comes in, a flashing banner appears on the web page with an underlying link (this works so far) * when the user klicks on this flashing banner, the external SIP UA which is already ringing, shall pick up the call. I know that it's possible to autoanswer a call with the intercom feature. Also the SIP client X-Lite which we use here is able to autoanswer a call. I however want to manually decide when the UA takes the call with the following workflow: * X-Lite rings on incoming call * user klicks on the flashing banner * X-Lite takes the call What is the best way to have this done? Move the call to park and then retransfer again with intercom, or is there a better solution? Best regards Peter

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outbound error log
by selva kumar 17 Jun '09

17 Jun '09

Hi Michael, I have pasted the freeswitch logs as requested in ( pastebin.freeswitch.org) Thanks Sam

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