- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Grandstream Freeze on Originate command
by rex.alex 07 Feb '11

07 Feb '11

Hello, This solution worked for me as well. Thank you so much. Rex -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Grandstream-Freeze-on-Origina… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Using mod_opal with MSN Messenger
by mazilo 07 Feb '11

07 Feb '11

Has anyone managed to configure mod_opal to register to MSN Messenger server to place/receive calls to/from any MSN Messenger users? I took a look at the conf/autoload_configs/opal.conf.xml file and don't know what to fill the gk-address, gk-identifier, and gk-interface. Anyone? ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Using-mod-opal-with-MSN-Messe… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Lua Session creation is blocking?
by Stephen Wilde 07 Feb '11

07 Feb '11

I have a lua script that originate a new call creating a session as: outbound_session = freeswitch.Session(dialstring, session); The problem I have is that session creation sometimes is blocking and sometimes is not blocking. To me it seems that is related to the RTP so the session creation is blocked until early media is present or until the called answer the call. Is this true or I have miss something? There is a way to do a non blocking session creation? Stephen

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blocking 183 w/o sdp
by Sam 07 Feb '11

07 Feb '11

Hi, how can i ignore 183 without sdp, what happens is the provider sends 183 without sdp and by applying "<action application="set" data="sip_ignore_183nosdp=true"/>" the FS sends 180 to the leg a. Here i want to block the 183 with SDP just like router as b2bua and send nothing to leg a, and when actual 183 with sdp comes it should send . Its because, providers are sending false signaling by sending 183 without sdp,and it hampers while @ production, Although by cisco sbc i have done this but i want to do it by FS, Take a scenario, when call is send 183 without sdp for 10 secs and then followed by 183 with sdp ( actual signal), but when some one dials invalid number it rings for 10 secs and then gives SIP cause 404, which is bad from the providers. So this is the reason i want to block it. Most of the providers do this, the way out is blocking. I have got an advice from Tihomir to do "execute_on_ring and parse your 180 / 183 messages in search of SDP, once you get 183 without SDP do not send it back to leg a and send signal only when you got 183 with sdp or 180 " And this could be valid call flow. This happens in many cases where the provider is having nextone as a sbc and that too tier 1 ! Regards Sam

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mod_h323 error
by Gustavo Espeche 07 Feb '11

07 Feb '11

Hello we are having a error with mod_h323 2011-02-06 22:04:06.999851 [ERR] mod_h323.cpp:2150 AUDIO RTP REPORTS ERROR: [Missing local port] when this error appear not more calls connect. someone have this error with mod_h323. regards. Gustavo Espeche www.easyipcall.com

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FSClient Release - A FreeSWITCH SIP Client for Windows in .NET
by Mitch Capper 06 Feb '11

06 Feb '11

In prep for tomorrows conference I am happy to release FSClient. I would like to thank DRK and especially jlenk for their continued support and help on this project. jlenk created the installer, fixed code, and certainly helped bring it along quite a bit faster! FSClient is a windows SIP client: Most features of any standard sip client multiple calls at once transfer, holding, speakerphone, DND multiple SIP accounts advanced headset support (caller id, buttons, etc) for jabra and plantronics out of the box (with plugin support for easily adding others) basic contact book support (with a sample XML contact book plugin provided) All codecs (minus commercial g729 support) that freeswitch supports Give it a shot and let us know about any issues you run into, overall it was decently tested for many months but some of the rapid changes as of late means there may be some larger bugs that crept in. I will also be on the conference call tomorrow and can answer questions there. And so to download the binaries: http://files.freeswitch.org/windows/installer/x86/FSClient.zip There will be a readme in the install dir on usage. The source code is available on the contrib git repo mitchcapper/FSClient. As for building from source you will need to do a few things to trunk currently. In the source folder you need to move mod_portaudio.c to the freeswitch\src\mod\endpoints\mod_portaudio folder ( http://jira.freeswitch.org/browse/FS-3006 are the changes pending trunk). In addition move the portaudio.2010.vxcproj over the version in freeswitch\libs\portaudio\build\msvc to enable direct X support. Make sure you have the WPF Toolkit also installed. Build trunk and then for FSClient set the ENV var FREESWITCH_SRC_LOCATION to the src location and it will also auto-copy the needed files into the build folder. History: Back in May I wanted a SIP client that supported the Jabra headsets, after not finding one that would work as I liked I looked to open source and everyone pointed me towards FSComm. FSComm is great, and has a huge amount of time poured into it, but the C++/QT base for it was not for me. .NET was the natural choice due to its rapid development time and stability so I went there. After a few weeks work I had a client that I used for over 6 months and was fairly stable with relatively minor changes. A few weeks ago on the conference call someone mentioned wanting a windows sip client option and I figured I would offer up my code base. Well getting private code ready for public use was a bit more complex than planned but its here now and certainly in the past several weeks has had a lot of development. ~Mitch

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Freeswitch and mysql
by Antonio 06 Feb '11

06 Feb '11

Hello, I asked and more I searched, I found nothing, how to integrate FreeSwitch with mysql. Some time ago I found Asterisk-realtime and wanted to know if there is something similar in FreeSwitch. Is to avoid double configurations, for example if there is a conference room that can be accessed from multiple servers, etc. I'm new in FreeSwitch and I have much knowledge ;-) Does anyone have any url or book on how to do? Lua?. Thanks. Greetings

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Re: [Freeswitch-users] play_and_get_digits question
by Madovsky 06 Feb '11

06 Feb '11

Ok I think I misunderstood the concept of var where digits should be put in. it's with this var I need to ckeck after this application. self forum thank you ;) ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 12:29 PM Subject: play_and_get_digits question I have this lines in my dialplan <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/> <action application="transfer" data="otherext XML features"/> if I don't press any DTMF, after 4 loops the dialplan continues, but I thought that it was automatically hangup in case of no digits bad digits... Did I forget something ? Thanks Franck

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play_and_get_digits question
by Madovsky 06 Feb '11

06 Feb '11

I have this lines in my dialplan <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/> <action application="transfer" data="otherext XML features"/> if I don't press any DTMF, after 4 loops the dialplan continues, but I thought that it was automatically hangup in case of no digits bad digits... Did I forget something ? Thanks Franck

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H.323 implementation on FS
by Vedran Zeljeznak 06 Feb '11

06 Feb '11

hi everyone, do any of you have any suggestions which H.323 implementation (mod_opal or mod_h323) should be used on a production Freeswitch platform (HEAD version)? Is it better to forward calls from H.323 trunk through Yate (configured as H.323 to SIP proxy) before terminating them on Freeswitch? --- Vedran Zeljeznak

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