- FreeSWITCH-dev - FreeSWITCH Mailing Lists

FreeSWITCH Conference Call Today: Repro SIP proxy
by Michael Collins 19 Sep '12

19 Sep '12

Hello all, Please join us for the conference call at 1PM EDT, 10AM PDT: http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_19 Scott and Daniel will be discussing the Repro SIP proxy and the ReSIProcate SIP stack. Talk to you soon. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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bv-tapi jira
by François Delawarde 19 Sep '12

19 Sep '12

Hi all, Now that bv-tapi is hosted on git.freeswitch.org, will there be a jira setup so we can send patches? If not, where should I send them? Thanks, François.

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FreeSWITCH Weekly News and Notes
by Michael Collins 17 Sep '12

17 Sep '12

Hello all, It's been another productive week on the FreeSWITCH team. We are pleased to let you know that we have officially tagged <http://freeswitch.org/node/418>FreeSWITCH version 1.2.2 in the git repo. Source tarballs are available in the usual location <http://files.freeswitch.org>. Thanks to all those whose efforts make more frequent releases a reality. It is much appreciated. On last Wednesday's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_12>we enjoyed a nice Adhearsion <http://adhearsion.com/> presentation by Ben Langfeld and Ben Klang. Adhearsion is a Ruby-based framework for developing telephony applications. Ben and Ben discuss how Adhearsion works, why Ruby is cool for building telephony apps, and why the Adhearsion guys love FreeSWITCH. FreeSWITCH community members are invited to join the Adhearsion team at AdhearsionConf <http://adhearsionconf.com> in Palo Alto, CA on October 20-21, 2012. Community members receive a special rate by using discount code AHNLOVESFREESWITCH. Thanks to Ben and Ben for a great presentation with cool slides<http://wiki.freeswitch.org/images/8/8e/FreeSWITCH_Weekly_CC_2012-09-12.pdf> . For the next few weeks we look forward to hearing from Daniel Pocock and Scott Godin who will be telling us more about the Repro<http://www.resiprocate.org/About_Repro>SIP proxy and the ReSIProcate <http://www.resiprocate.org/Resip_Overview> SIP stack. For many of us it will be our first look at a SIP proxy that does not have its roots in the OpenSER project. We look forward to learning more on this Wednesday's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_19> . Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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FreeSWITCH Conference Call Today
by Michael Collins 12 Sep '12

12 Sep '12

Hello All, Our FreeSWITCH conference call will be taking place in a bit. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_12 We look forward to hearing from Ben and Ben about the Adhearsion project. We also have a few community updates. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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FreeSWITCH Weekly News and Notes
by Michael Collins 10 Sep '12

10 Sep '12

Greetings! We hope you all had a great week. On our Wednesday conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_05>we discussed several items as a community. One item of note was how to handle mailing list posts that are overly broad and reflect a lack of research on the part of the individual doing the posting. After much discussion we decided that we would create some online documentation that helps new ones get their bearings when considering the big picture in FreeSWITCH. (For example, what are modules and why do we have them?) Thanks to Dave Kompel for helping to get that started. We are also pleased to announce that we have started up the Adopt-a-module<http://wiki.freeswitch.org/wiki/Adopt-a-module>project. The idea is simple but powerful: community members who are knowledgeable and enthusiastic about a specific module will volunteer to "adopt" that module. Adopting a module means doing several things: watching the mailing list and IRC channel for questions, monitoring the Git repository for new commits, keeping the module's wiki page up-to-date, and acting as a bug marshal for any Jira <http://jira.freeswitch.org> tickets that are opened. We've had several people step up already. Please visit the list of modules needing adoption<http://wiki.freeswitch.org/wiki/Adopt-a-module#Modules_Waiting_for_Adoption>to see if you there is one that fits your area of expertise. We give a special note of thanks to Anshel Blum for helping to get this one going. This week we welcome Ben Langfield and Ben Klang<http://www.cluecon.com/speaker/111/>who will be discussing the Adhearsion <http://adhearsion.com/> framework for FreeSWITCH. Adhearsion is a Ruby-based framework for building telephony applications. You may recall that Ben Klang joined us at ClueCon 2012 <http://www.cluecon.com> to make the announcement<http://www.cluecon.com/presentation/news-about-adhearsion-project/>about Adhearsion being available for FreeSWITCH. We look forward to learning more about how Adhearsion works with FreeSWITCH. Let's all have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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FreeSWITCH Conference Call Today
by Michael Collins 05 Sep '12

05 Sep '12

Hello all! Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_05 We have a number of things to discuss as a community. Look forward to seeing you there. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Conference call without media
by Papineni, Suneel 04 Sep '12

04 Sep '12

Hi, I am having a specific requirement with respect to conference. I want to transfer a call to an existing conference ignoring media or codec parameters of that call leg. Is there any channel-variable that can be set for this call leg or during transfer so that call is joined to conference even without passing any media or codec information. I can resolve my situation even if there is a way to ignore media transmission (details) when first conference call is created. Thanks & Regards Suneel

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FreeSWITCH Weekly News and Notes
by Michael Collins 04 Sep '12

04 Sep '12

Happy Tuesday to you all! I hope you had a relaxing 3-day weekend. On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_29> we had a nice discussion about TLS <http://wiki.freeswitch.org/wiki/Tls> in FreeSWITCH. Mitch Capper helped demystify some of the things that go on with creating certificates, handling certificate authorities (CA), and the like. This week we will be having a community discussion on how to improve our handling of poorly-worded or inappropriate questions sent to the mailing list. Ourupcoming schedule<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_29#Upcoming_Presentations> for September is all booked up, so make plans to be on our Wednesday conference calls. Be on the lookout for updated English Callie and French June sounds. They have been recorded and are being verified. I suspect they will show up in the next day or two. The new FreeSWITCH book is moving along as well. I just submitted chapter 2 to our publisher and am now on to redrafting chapter 3. On a side note I just want to mention that I updated the FreeSWITCH Wikipedia page<http://en.wikipedia.org/wiki/FreeSWITCH> to reflect the fact that we are now on version 1.2.1 and that we've formed stable and development branches. ClueCon 2012 attendees and FreeSWITCH community members qualify for special pricing at the IIT Real-Time Communications Conference and Expo<http://www.rtc-conference.com/register/>. Use the special discount code *FREWDSC* when you register. More details are available at ClueCon.com <http://cluecon.com/>. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Bypass media from dingling to SIP
by Richard Screene 03 Sep '12

03 Sep '12

Hello, I have a FreeSWITCH that bridges a mod_dingaling call to a SIP call. This is working well. But, I am now trying to use the bypass media or proxy media flags and I am having less success! (It would appear that there is no SDP present in the outgoing SIP INVITE.) Here is the dialplan I'm using: <include> <extension name="jingle_bridge"> <condition field="destination_number" expression="^1000$" /> <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/external/abc@<IP>:5060"/> </condition> </extension> </include> Should it possible to use these flags between Jingle and SIP? Many thanks, Richard Richard Screene Senior Developer NetDev Ltd & Drum Collaboration +44 1273 936125 www.netdev.co.uk www.thisisdrum.com Registered in England and Wales Company Number 04741258

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question on file format codec
by Seven Du 01 Sep '12

01 Sep '12

Hi, If you noticed the video-media-bug branch, I managed to record a video file with the following methods conference 3000 record /tmp/testrecord.fsv uuid_record <uuid> start /tmp/testrecord.fsv The first command works perfect as conference voice is L16, but the last one has bad audio I guess because it's PCMU ?. I noticed a SWITCH_FILE_NATIVE flags which is supposed to disable transcoding but in my case in mod_fsv the flag is not set so should it transcode to L16 before writing to the fsv file? Thanks. -- Seven Du Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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