December 2010 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] ftmod_libpri - Channel Selection patch
by lakshmanan ganapathy 11 Dec '10

11 Dec '10

Hi all, When using ftmod_libpri, if a incoming call comes without "Channel Identification" ( in SETUP packet ), then the call didn't get proceeded. I've taken some code snippets from ftmod_isdn and I applied it in ftmod_libpri. I've tested it and in my setup it works fine. I've attached the patch here. Please verify it and found ok commit it. regards, Lakshmanan G.

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[Freeswitch-dev] Scriptable test-suite
by Joegen E. Baclor 10 Dec '10

10 Dec '10

Hi, We are trying to work out a test suite for sipXecs media services and we thought that freeswitch itself is the best place to do it. I am initiallly thinking of controlling freeswitch using ESL inbound to simulate traversing of sipXecs media services ivr menus. However, although ESL has the most flexibility among the choices, I find the scripting modules really appealing specially because it exposes the ability to add new test scenarios. Am I correct that ESL inbound function(eg: placing independent outbound calls without having to hit a dialplan) is not currently doable via the mod_scripting_language plugins? Because of this, I am plannning to simply expose ESL inbound/outbound library via a google::v8 glue to make the test scenarios somehow scriptable. I am wondering if I am not crazy if i do so. I have came accross this thread old http://thread.gmane.org/gmane.comp.telephony.freeswitch.user/10926/focus=10… . I am wondeing if more work/thoughts came into this so i wont be reinventing the wheel. advice appreciated. Thanks! Joegen

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[Freeswitch-dev] Asterisk migrations
by Michael Collins 10 Dec '10

10 Dec '10

Hey gang, I was just curious if anyone out there had created any tools to assist with migrating from Asterisk (or Trixbox, etc.) to FreeSWITCH. We're starting to see this scenario more and I would like to find out how our community members are handling it. Thanks, MC

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[Freeswitch-dev] User Interface Programmer Job Opportunity
by Anthony Minessale 09 Dec '10

09 Dec '10

If anyone who lives in the US especially [MI WI OK CA] or would like to move to one of those places and is good at UI design. Contact jobs(a)freeswitch.org with a resume. We need some programmers to work with us on the CudaTel UI. Its a demanding yet rewarding job in software development and you get to help with FreeSWITCH as well. Jquery/Js/Comet type skills are the most important. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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[Freeswitch-dev] FreeSWITCH Conference Call Today
by Michael Collins 08 Dec '10

08 Dec '10

Don't forget about today's call! http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_08 Our agenda is light today so bring your questions and suggestions and we'll discuss them as a group. -MC

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[Freeswitch-dev] OSTAG - How You Can Help
by Michael Collins 08 Dec '10

08 Dec '10

Greetings! The OSTAG team would like to let everyone know that we are ready for non-profit work. While monetary donations are always welcomed they certainly are not the only way that you can help. We have added a new section to the ostag.org Web site on how to donate <http://www.ostag.org/node/7>. Please review it and think about how you can donate time, money, or resources to the OSTAG project. All donations are tax-deductible. Thank you for continuing to support Open Source software! -The OSTAG Team

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[Freeswitch-dev] playback on uuid_bridge
by Kevin Raison 06 Dec '10

06 Dec '10

I would like to play a wav file on only one leg of a bridged call (setup using uuid_bridge) in order to keep the message private to that leg. Using "playback" plays the message on both legs. Is there a way to make the playback private? Thanks, Kevin Raison

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[Freeswitch-dev] mod_khomp
by Geovani Ricardo Wiedenhoft 06 Dec '10

06 Dec '10

The Endpoint and the documentation for the Khomp boards (mod_khomp) are available on the official FreeSWTICH branch. Also, the documents are available on the wiki. http://wiki.freeswitch.org/wiki/Khomp The version is compatible with all Khomp boards (SPX series): - FXS - FXO - E1 - R2, R2 DTMF and OpenCAS in Hardware, ISDN (User, Network), OpenCCS, LineSide, CAS_EL7, E1LC - GSM with send/receive SMS (boards and usb devices) - Passive record (R2, ISDN, OpenCAS, OpenCCS and FXO) - kommuter Visit our site to more information about Khomp products: http://www.khomp.com.br Thank you. :) Khomp development team

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[Freeswitch-dev] Bridged call duration
by Dmitry Mordovin 06 Dec '10

06 Dec '10

Hello All! Can't find 'bridge' options to set maximum call duration. There is a way to do that? Best regards, Dmitry Mordovin

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[Freeswitch-dev] FreeSWITCH module controlled by external application
by Alexander Lyasin 03 Dec '10

03 Dec '10

Hello all! I interests the next questions: Are Possible to use FreeWSITCH how UA? I have a task to write FreeSWITCH module (for Linux) that get an information (a signals) from another application and establishes SIP transactions (with FreeSWITCH functions) with another SIP agents. Ex: An Application sends a signal to the FreeSWITCH module (eg using a message queue), transmits the data about the caller and the called party. The module should to establish the session between them with FreeSWITCH functions. In this case voice will not be transmitted with FreeSWITCH. Perhaps this problem has been resolved already. Can are you provide me a link to a similar case, examples or documentation, how it done within FreeSWITCH (preferably a C )? Thank you very much!

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