- FreeSWITCH-dev - FreeSWITCH Mailing Lists

ESL, Swig and C#
by Christian Jensen 03 Sep '09

03 Sep '09

I am about to embark on building a patch that will build a C# version of ESL using Swig What is the preference - hardcoding the rebuild into the VS project, nmake, .cmd or .bat? The biggest issue will be the location of swig - either that goes into the environment vars or I point to a location on my drive. Would it be possible to include it in a download or something like the other libs? Any help in direction is very welcome. Thanks!

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start_dtmf bug?
by Trixter aka Bret McDanel 03 Sep '09

03 Sep '09

I am trying to use portaudio to bride a radio device into freeswitch. I encountered a problem that I believe to be a bug, although I am not sure, so I am asking if anyone can confirm this. <extension name="radio_conference"> <condition field="destination_number" expression="^123$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="start_dtmf"/> <action application="conference" data="radio@default"/> </condition> </extension> if I call that from portaudio all input from the microphone appears to be discarded. If I call it without the start_dtmf it works as expected. To avoid any potential for confusion, I do actually need inband DTMF detection from the radio side of things. I only looked at this from the portaudio side of things, so if anyone who has an otherwise working portaudio configuration could look to see if this is present in their systems I would appreciate it to know if this is something I did or something else. Once confirmed I will look at posting a bug report, but I dont want to do it if its just me. Thanks, -- Trixter http://www.0xdecafbad.com Bret McDanel pgp key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x8AE5C721

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Scalabilty of Freeswitch
by Jerry Richards 02 Sep '09

02 Sep '09

Hello All, Does anyone know what the capacity of a stand-alone Freeswitch (assuming a top-of-the-line server), in terms of how many users? Also, when that number is exceeded, how can Freeswitch server be distributed to accommodate a larger installation? Best Regards, Jerry

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ANNOUNCEMENT: Friday Public Meetings Are Coming Back!
by Michael Collins 01 Sep '09

01 Sep '09

We are happy to announce <http://www.freeswitch.org/node/201> that the Friday public FreeSWITCH meetings are returning, starting this Friday, September 4. Meetings will run from 11am to 5pm CST. The meetings will be held in the FreeSWITCH public conference, also known as the 888 conference. Connection options include: * SIP: 888(a)conference.freeswitch.org * IAX: 888(a)conference.freeswitch.org * H.323: 888(a)conference.freeswitch.org * GoogleTalk: 888(a)conference.freeswitch.org * PSTN: 1-213-799-1400 Please join us and be a part of the conversation! We will be discussing agenda items that include programming, documentation, and janitorial projects. We welcome your input. Please bring your questions, suggestions, and ideas. If you have specific ideas for an agenda item that you feel should be discussed then please email myself and Brian West off list. We will post the agenda for each meeting on the FreeSWITCH wiki page. Thanks for helping make FreeSWITCH such a great community! -Michael

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video support in FS
by Erwin Huang 31 Aug '09

31 Aug '09

Hi, I wonder if there is any sample application for mod_fsv. What video codec is supported? Thanks, Regards,

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Fwd: FS + G729
by mauro＠nohad.com.br 31 Aug '09

31 Aug '09

Hi there Interesting point. We have licenses for 723 and 729 for asterisk, and we also need to use asterisk distribution to talk with our voice card for PSTN. We dont have the drives for this card developed for freeswitch yet. So, for us the freeswitch is a great solution, and were beginning to use it, but for now we also must use asterisk for some things that: or we dont know how to do yet, or we feel more comfortable with, or we have to. I think that this is a conservative and secure approach for us, since we just beginning to scratch freeswitch. We tried to do exactly what is described in the message (bottom) but we have no success. Like I said we use and must to use asterisk for now, and we already have the 729 and 723 licenses. I know that 729 (and 723?) license like asterisk is on the way, and also know that there is at least one vendor for 729 for freeswitch, but it will be great if someone could give us the direction to make a extension registered in freeswitch to talk with another thought asterisk to make the codec trans-coding in asterisk. I also support the idea to solve things like this in the freeswitch project (were not talking here about asterisk) but for now (for us) to make a good use of the 923459283752938 good things in freeswitch we must to use it together with asterisk. Any idea to this direction (transcoding thought asterisk) would be great appreciated. Regards, Mauro ---------- Forwarded message ---------- From: Michael Collins < <mailto:msc@freeswitch.org> msc(a)freeswitch.org> Date: Fri, Aug 28, 2009 at 8:36 PM Subject: Re: [Freeswitch-dev] FS + G729 To: <mailto:freeswitch-dev@lists.freeswitch.org> freeswitch-dev(a)lists.freeswitch.org On Fri, Aug 28, 2009 at 3:20 PM, Andre Mendes . < <mailto:andremendes2004@gmail.com> andremendes2004(a)gmail.com> wrote: Hi, I want to use G729 like Asterisk G729 in Freeswitch, any source to make this codec work with FS? I make a install asterisk with G729, but I have a problem: FS - end point 1000 - G711 make a call to end point 1001 - G729 use a asterisk trunk to make a transcoding codec. Not work yet. Please help to make this only in FS or using a Asterisk to make a transcoding. André Mendes FreeSWITCH can do G729 is "passthrough" or proxy mode as well as bypass media mode. However, no transcoding yet. It's in the works. -MC _______________________________________________ FreeSWITCH-dev mailing list <mailto:FreeSWITCH-dev@lists.freeswitch.org> FreeSWITCH-dev(a)lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-dev> http://lists.freeswitch.org/mailman/options/freeswitch-dev <http://www.freeswitch.org> http://www.freeswitch.org

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Question about mod_limit
by Anatoliy Kounitskiy 31 Aug '09

31 Aug '09

I have question about mod_limit: is there a reason to have difference between the api applications and the dialplan? I mean when you use db function from the cli you can use insert/select/delete but when you use it from the dialplan you can only use insert/delete. #define DB_USAGE "[insert|delete]/<realm>/<key>/<val>" #define DB_DESC "save data" SWITCH_STANDARD_APP(db_function) but in the SWITCH_STANDARD_API(db_api_function) i have three options insert,delete and select What i try to accomplish is to be able to insert values from the dialplan and after that to be able to use them ( to select and use the result) from another extension. Thank you in advanced, -- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy(a)kounitskiy.com Mobile: +359898913540

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Portaudio upgrade
by Tamas 29 Aug '09

29 Aug '09

Hello, I'm trying to upgrade the portaudio lib due to some issues with the sound card in my box (I've found issues on 2 different boxes). I'm trying to run FS on bleeding edge distro - as this is a desktop box requiring fancy things - Ubuntu Jaunty 9.04 x86_64/i386 I know that hot new ubuntu is not supported so I'm trying to work on by myself - and give back if I have anything reasonable :) As I saw on portaudio.com, there were some improvements with ALSA handling and it should handle newer cards better now (non-mmap patch): http://portaudio.com/trac/log/portaudio/trunk/src/hostapi/alsa/pa_linux_als… As I've seen, there is some fix for windows's wdmks stuff too. I'm not qualified to hack PA, but I can play with it a bit :) What I would need to know is, which version (and source - as there is a separate fork e.g. at audacity too) is the one the FS trunk is having and what fixes/changes had been applied to it before merge (if I have had the exact version of included PA, I could make a diff and see by myself). From FS svn log I can see what changes were since the version bump- From FS svn log I see, that at r8835 we had the last version bump: r8835 | anthm | 2008-06-25 00:15:50 +0200 (sze, 25 jún 2008) | 1 line merge in newer portaudio Are the changes in FS regarding PA sent to upstream? (I know, PA is used to be pretty non-responsive) Maybe it would be good to store the last PA version and source data in tree to be able to upgrade easier in the future. We are interested in improving the quality (mainly on windows) as we try to use this excellent stuff as a softphone base :) Regards, Tamas

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FS + G729
by Andre Mendes . 29 Aug '09

29 Aug '09

Hi, I want to use G729 like Asterisk G729 in Freeswitch, any source to make this codec work with FS? I make a install asterisk with G729, but I have a problem: FS - end point 1000 - G711 make a call to end point 1001 - G729 use a asterisk trunk to make a transcoding codec. Not work yet. Please help to make this only in FS or using a Asterisk to make a transcoding. André Mendes

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Unable to access directory variable from channel
by Juan Backson 27 Aug '09

27 Aug '09

Hi, I set up a variable called "account-id" in the user directory xml file. >From the info app, I can't find the variable for variable_account-id. Also, in my C module, I tried to obtain the value with switch_channel_get_variable(channel,"account-id") and it is not returning null. Does anyone know how I can access the user variable from within my mod? Thanks, JB

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