January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Siptapi and Freeswitch
by Peter P GMX 10 Sep '10

10 Sep '10

Anybody tried siptapi with freeswitch? http://sourceforge.net/projects/siptapi/ This may enable Click2Dial e.g. from Outlook to Freeswitch. So anybody has experience with that solution? Best regards Peter

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error loading module 'luasql.mysql'
by Adam Wilt 02 Sep '10

02 Sep '10

Hi, I followed the instructions in the Lua documentation for setting up luasql, but when I try to run my script I get: 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so': /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment prot after reloc: Permission denied stack traceback: [C]: ? [C]: in function 'require' /usr/local/freeswitch/scripts/l.lua:2: in main chunk I'm running FreeSWITCH version 1.4 and luasql version 2.1.1. I changed the permissions for mysql.so and for my script to 777, so I'm not sure where the permission problem could be. I'd appreciate any suggestions. Thanks, Adam

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Stream audio file/live to multiple SIP endpoints with IP multicast
by Sluschny, Thomas 30 Aug '10

30 Aug '10

Hi, i want to stream audio (from file and live) from FreeSwitch to multiple SIP endpoints (these are proprietary). Because of bandwith limitations we have to use IP multicast for audio data. Now the question: how can i do this smartly in FreeSwitch? I have read some Wiki entries: - mod_esf: seams to use multicast, but i couldn't find usage information - mod_shout: streaming, but from internet ? MP3, but i want wav-files to stream - mod_local_stream: seams to play files in a loop, i want to play a whole file once. - mod_conference: seams not to support IP multicast Im searching for a simple way to send a SIP INVITE to all my endpoints and then stream audio (live from soundcard or from file) over IP multicast to all these endpoints. Thanks in advance, Thomas

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[contrib] a new java ESL inbound/outbound client
by david varnes 30 Aug '10

30 Aug '10

Hi all, I had a basic java inbound ESL client kicking around that I have used in a couple of small projects over the last year. I needed something a little more complete for a new project so I dusted it off and made it less incomplete. It still needs more work and testing, but it certainly is usable right now. I have tested it against FS 1.0.4 and latest trunk. I have put it in my contrib area in svn in hopes that some may find it useful: http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/dvarnes/java/esl-cli… I would be interested in any feedback ... Features * Apache License (ASL) version 2 * Standalone Inbound client * Framework classes to easily create an Outbound socket client * based on Netty [1] nio library version 3.1.5.GA (previously was using Apache MINA, but this is easier) * logging via slf4j * only dependencies are slf4j-api and netty (both Apache licensed) * single jar which is a valid OSGi bundle * built using maven * eclipse projects * reasonable level of java docs Still todo * Docs * Simple example apps * .. more in TODO.txt in project root. There is no binary jar available right now since I don't know how/if I can put files up to file.freeswitch.org. In the meantime to build you need maven [2] installed. If you are unfamiliar with maven usage, I can post a simple howto. davidv [1] http://www.jboss.org/netty/downloads.html [2] http://maven.apache.org

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Direct inward dialling
by Peter P GMX 07 Jun '10

07 Jun '10

Dear all, when I get an incoming call from my SIP provider I do receive an invite on the trunk number, e.g. in Germany 0xxxxx9. However I have an extension block of 0xxxxx90 to 0xxxxx999. In the dialplan I checked the condition field="destination_number" but this compares to the number of the trunk 0xxxxx9, so I do not get the direct inward dialling extensions. The SIP to-header contains the direct inward dialling number 0xxxxx90 to 0xxxxx999 but I have no success in accessing it in the dialplan. I tried sip_to_user, sip_to_uri, sip_req_user, sip_req_uri, but no success. They are all empty. Any idea which variable to parse? Best regards Peter

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Re: [Freeswitch-users] Grandstream gateways
by Samuel Mukoti 23 Apr '10

23 Apr '10

Thank you for those tips, I do have some small setups using gxw4108 they work or, except CID doesn't seem to work. I will try the channel bank route - just don't know too much about the setup options or how you'd purchase the correct config, eg. For 150 FXS channel bank, can I get a single PCI card for that? I may end up using the grandstream fxs gateways then use the T1 channel bank from sangoma, Thank you all.. Lastly, I know asterisk now has an offical skype_ module, Is there anything similar I could use? On 25 Nov,2009, at 9:52 PM, Cory Andrews <cory(a)voipsupply.com> wrote: > Samuel - you could go with FXS gateways or channel banks. If you go > the gateway route Grandstream or Audiocodes would work fine. > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > you could use a 24FXO channel bank cross connected with Rhino T1 > cards, or individual FXO gateways but you may have a hard time > finding 24 ports of FXO in a single GW. Best performing T1 cards in > my experience (thousands of deployments) are Sangoma. Your server > configuration looks fine. > > Cory J. Andrews > Director New Market Initiatives > > Sayers Media Group > VoIP Supply, LLC > 454 Sonwil Drive > Buffalo, NY 14225 > 716-250-3402 OFFICE > 716-630-1548 FAX > 716-601-4474 MOBILE > candrews(a)sayersmedia.com > > > Have I exceeded your expectations? Please share your experience > with my boss, Benjamin P. Sayers, CEO > > NOTICE: The information contained in this email and any document > attached hereto is intended only for the named recipient(s). It is > the property of the VoIP Supply, LLC and shall not be used, > disclosed or reproduced without the express written consent of VoIP > Supply, LLC. If you are not the intended recipient, nor the employee > or agent responsible for delivering this message in confidence to > the intended recipient(s), you are hereby notified that you have > received this transmittal in error, and any review, dissemination, > distribution or copying of this transmittal or its attachments is > strictly prohibited. If you have received this transmittal and/or > attachments in error, please notify me immediately by reply e-mail > or telephone and then delete this message, including any > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > 14225 USA. > > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: [Freeswitch-users] Grandstream gateways > > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. I wanted assistance with choosing the most appropriate > hardware. We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request(a)lists.freeswitch.org > wrote: > >> 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: mod_conference kick to abort invitations (Michael Jerris) >> 2. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Michael Jerris) >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> 4. Re: remote_media_ip variable not set (Michael Jerris) >> 5. Re: How to find whether the destination extension supports >> encryption (Michael Jerris) >> 6. Re: Bypass_media and re_invite (srinivasula reddy) >> 7. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Stephen Crosby) >> 8. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Tihomir Culjaga) >> >> >> --- >> ------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> From: Michael Jerris <mike(a)jerris.com> >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> invitations >> To: freeswitch-users(a)lists.freeswitch.org >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B(a)jerris.com> >> Content-Type: text/plain; charset="windows-1252" >> >> Its a feature we don't have, patches welcome. >> >> Mike >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >>> Hi members, >>> I?m controlling freeswitch with the conference module via xmlrpc. >>> >>> Is it desired that the kick command can only kick users that are >>> connected to the conference? >>> Is there no chance abort an invitation? >>> The kick command has no effect until the person I invited with the >>> dial command is connected. >> >>

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SIP Overlap support?
by Dennis 06 Apr '10

06 Apr '10

hi there, i would like to ask, if fs has support for something like "SIP Overlap"? instead of receiving the phonenumber from our carrier in a block, we want to receive the phonenumber digit-by-digit and we want to tell fs when the number is complete. our carrier could send us the phonenumber digit-by-digit, but what about the fs-side? thanks and kind regards dennis

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compilation error of skypiax_protocol.c
by Jingwei Yang 25 Feb '10

25 Feb '10

Hi Guys, I got a compilation error of skypiax_protocol.c with the latest version r15764. Compiling skypiax_protocol.c... *cc1: warnings being treated as errors* skypiax_protocol.c: In function â€˜X11_errors_handlerâ€™: skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function â€˜skypiax_send_messageâ€™: skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function â€˜skypiax_do_skypeapi_thread_funcâ€™: skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and code make[5]: *** [skypiax_protocol.o] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_skypiax-install] Error 1 make[2]: *** [install-recursive] Error 1 I personally checked the file and it shouldn't be a merge problem. Does anyone encounter this as well?

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Re: [Freeswitch-users] Establishing a Call from .Net based application
by Diego Toro 19 Feb '10

19 Feb '10

Hi, the answer is yes, you can to use mod_managed wich offer C# managed class to call control http://wiki.freeswitch.org/wiki/Mod_managed. Or using managed ESL (libs/esl/managed) which offer C# managed class to receive and send events and commands to FreeSwitch. Diego Toro http://lacarretade.blogspot.com/ --- On Wed, 1/20/10, Scott Fernandez <scottferri09(a)gmail.com> wrote: > From: Scott Fernandez <scottferri09(a)gmail.com> > Subject: Re: [Freeswitch-users] Establishing a Call from .Net based application > To: freeswitch-users(a)lists.freeswitch.org > Date: Wednesday, January 20, 2010, 2:17 AM > Thanks Dome. Will try it out and get back to > you if I come across any issues. > > Regards, > Scott. > > On Wed, Jan 20, 2010 at 11:02 AM, > Dome Charoenyost <dome(a)tel.co.th> > wrote: > > Please try http://wiki.freeswitch.org/wiki/Webapi > > > you can create class and map to webapi. > > > > Dome C. > > > > 2010/1/19 Scott Fernandez <scottferri09(a)gmail.com>: > > > Hi, > > > > > > Is there any API modules available for me to initiate > a call from .Net based > > > application?. > > > > > > The idea is to include the API modules if any with the > .NET base classes so > > > that the API commands will be made available on it. I > know it is doable when > > > I use socket programming in .NET in which Telnet > session is created. > > > However, this would potentially hamper the performance > of the application > > > because of multiple sessions that will be created for > each call. > > > > > > Other than that, Is there any Freeswitch API modules > (like plug-ins) > > > available in order to include it into the .Net classes > and start building > > > the customized application? > > > > > > Any help from any one is highly appreciated. > > > > > > Thanks, > > > Scott > > > > > > > _______________________________________________ > > > 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 > > > > > > > > > > _______________________________________________ > > 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 > > > > > -----Inline Attachment Follows----- > > _______________________________________________ > 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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Announcement: FreeSWITCH Add Broadsoft SCA Support
by Michael Collins 17 Feb '10

17 Feb '10

We want to let everyone know that FreeSWITCH now supports the Broadsoft SCA method of doing shared lines. The story is here: http://www.freeswitch.org/node/227 Tony and Brian spent many hours laboring over this, so please be sure to show your appreciation to them for this new feature and all of the great things they do for the FreeSWITCH community and VoIP in general! -Michael

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