February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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${domain} not working
by Christian Hiller 06 Feb '11

06 Feb '11

Hello, from CLI > eval ${domain} is returning -ERR with latest git (FreeSWITCH Version 1.0.head (git-f60fdf6 2011-02-02 16-22-43 -0600)). Using this within a regexp is crashing FS. Kind regards Christian

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Auto test FS install - now necessary?
by Avi Marcus 04 Feb '11

04 Feb '11

Maybe I'm just being paranoid, but I've been seeing a lot of "I just updated to latest git and X broke" And that x is always something else... I'm not a seasoned coder on this magnitude, but perhaps we can develop a test suite (at least part of one) to test that a fresh install is functioning? A FULL test would probably be a huge undertaking and involve recqual for testing media, too... Also, each mod would need to have it's own test suite. Due to the rapid updating of FreeSWITCH this not be what you want to put your effort into, but... Thoughts? On Feb 4, 2011 4:04 AM, "Anirudha Shimpi" <anu(a)familytv.com> wrote: > Just downloaded (from git) and installed FS 1.0.7, however, every time I do > originate user/1001 &park, FS crashes. Running Centos 5.5 64 bit. >

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mod_directory multi-tenant
by Erick Baum 04 Feb '11

04 Feb '11

Is there a trick to getting the mod_directory application working in a multi-tenant configuration? I think I've tried every possible combination of settings and I cannot for the life of me get it to work. I can get the directory to answer but no matter what I put in to the search, it comes back with no results. Any help would be appreciated! Erick

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Sent aastra-xml event via esl?
by Tim St. Pierre 04 Feb '11

04 Feb '11

Hello, I'm trying to use ESL to send an event to aastra phones, so I can get them to execute an XML script and do other fancy things. To start, I'm just testing using a simple telnet connection to make sure it will work. I can successfully connect, and do things like "api help", etc. Here's what I have been sending: sendevent NOTIFY profile: internal event-string: aastra-xml user: 5101(a)communicatefreely.net host: stefan.151front.communicatefreely.net content-type: application/simple-message-summary I get this response: Content-Type: command/reply Reply-Text: +OK I don't see anything at all go out to the phone. Here's the phone I want to get the event: sofia status profile internal user 5101(a)communicatefreely.net Registrations: ================================================================================================= Call-ID: 996859d51795564d User: 5101(a)communicatefreely.net Contact: "Tim St. Pierre" <contact obscured> Agent: Aastra 6731i/2.5.2.30 Status: Registered(UDP-NAT)(unknown) EXP(2011-02-04 17:21:56) EXPSECS(1167) Host: stefan.151front.communicatefreely.net IP: User: 5101 Auth-Realm: pbx.communicatefreely.net MWI-Account: 5101(a)eccentricartists.communicatefreely.net Total items returned: 1 ================================================================================================= What is the syntax of the user: and host: headers? Is the host: the switch I want to respond to the event, or the IP address of the phone? I need FS to send the event to the phone based on it's registration. Thanks! -Tim

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Grandstream Freeze on Originate command
by rex.alex 04 Feb '11

04 Feb '11

Hi, I am using the api originate command to establish an internal extn to extn call (originate sofia/internal/2000%10.10.22.31 1000), where the 2000 registered in Grandstream GXP280 1.2.1.4 and 1000 registered in X-Lite. This command freezes the grandstream when I answer. I have pastebin the sip trace + freeswitch log in below mentioned url. Request your assistance in resolving this. http://pastebin.freeswitch.org/15222 Thanks, Regards, 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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