December 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

WebRTC Call
by Ros P 04 Dec '12

04 Dec '12

Hello, Has anyone had any luck in implementing a webrtc call? I have been able to register using the webrtc2sip gw and sipml5 client but no audio goes through. As this seems to be a client problem (issue 39<http://code.google.com/p/sipml5/issues/detail?id=39>), does anyone have any other suggestions? Anything successfully tested? Kind Regards, ros

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Re: [Freeswitch-users] voice announcement with Reason sip header
by Adam Kuśmirek 04 Dec '12

04 Dec '12

Hi, Url of the pb post: http://pastebin.freeswitch.org/20284 Thanks for any help Regards Adam

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RMPT module gets stuck!
by qasimakhan＠gmail.com 04 Dec '12

04 Dec '12

Hi, I am using RTMP module in freeswitch and after a few days one of the processors gets stuck on 100% processing. Here are a few logs that might help. freeswitch@internal> rtmp status profile default sessions > Profile: default > I/O Backend: tcp > Bind address: 0.0.0.0:1935 > Active calls: 0 > > Sessions: > uuid,address,user,domain,flashVer > 6094eee2-f9a2-427e-ada4-d4a9052525b0,X.X.X.X:54316,NNNNNNNNN,X.X.X.X,WIN > 11,5,31,2 > a1bb87a4-98a7-4b1e-b4d5-080e54690c31,X.X.X.X:50667,NNNNNNNNNN,X.X.X.X,WIN > 11,5,502,110 > f2df9e88-17b3-4685-9336-01fcdd2a5a0d,X.X.X.X:51806,NNNNNNNNNN,X.X.X.X,WIN > 11,5,502,110 > f9d46493-a707-48a5-9cbb-032aa80919bb,X.X.X.X:14331,,,WIN 11,5,502,110 > 7a561e8b-4d34-49ba-b4e8-b6c9ec58abd5,X.X.X.X:51998,,,WIN 11,4,402,287 > All the IP's are replaced by X.X.X.X and dialed numbers are replaced by NNNNNNN. My rtmp.conf.xml looks something like: > <configuration name="rtmp.conf" description="RTMP Endpoint"> > <profiles> > <profile name="default"> > <settings> > <param name="bind-address" value="0.0.0.0:1935" /> > <param name="context" value="public" /> > <param name="dialplan" value="XML" /> >  > <param name="auth-calls" value="true" /> >  > <param name="buffer-len" value="128" /> >  > <param name="chunksize" value="128" /> > </settings> > </profile> > </profiles> > </configuration> > I have tried several freeswitch versions but the problem persists. Please let me know if anything else is needed i will provide it. Regards, Qasim

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Questions about RTMP endpoint
by Germán Ruiz 04 Dec '12

04 Dec '12

Hi, I'm adding voice calls to a web site. I'm using RTMP endpoint and Flex clientprovided. I have the following questions: - Can flex clients (using RTMP protocol) register their presence and are visible by others connected using SIP? - Is it possible to send and receive text messages with RTMP protocol? Thanks Germán

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voice announcement with Reason sip header
by Adam Kuśmirek 04 Dec '12

04 Dec '12

Hi All I have Freeswitch connected to SIP Trunk from local Service Provider. Service Provider has NGN platform based on Broadworks applications. When making failed outbound connections I never get SIP responses with hangup cause, but they play 20 sec. voice announcement (eg. unallocated number ...) and send 487 Request Terminated containing Reason header after that. ---> INVITE <--- 100 Trying <--- 183 Session Progress <--- 183 Session Progress with header Reason: Q.850;cause=? <--- 20 sec. voice announcement <--- 487 Request Terminated with header Reason: Q.850;cause=? ---> ACK I'd like to hangup the call after 183 Session Progress containing header Reason: Q.850;cause=?. First i tried execute_on_media and execute_on_pre_answer, but the scripts are triggered only for the first 183 Session Progress that does not contain Reason header. Then I tried to resolve the problem by connecting to Outbound Socket, but I can see only one CHANNEL_PROGRESS_MEDIA event for the first 183 Session Progress. Can anyone help ? Regards Adam

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Genband S3 Intelligent SBC
by Gavin Henry 03 Dec '12

03 Dec '12

Hi all, It looks like as we order more and more capacity for our UK Ofcom number allocations that we host with BT Wholesale, we are going to be migrating to Genband S3 Intelligent SBC from the ACME packets we currently interconnect with using FreeSWITCH: http://www.genband.com/media-center/press-releases/bt-offers-innovative-glo… Anyone else using FreeSWITCH with the Genband S3 SBC? Issues? Recommendations? Thanks, Gavin. -- http://www.suretecsystems.com/services/openldap/ http://www.surevoip.co.uk

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mod_cdr_mongodb follows json rfc4627?
by Jose Fco. Irles Durá 03 Dec '12

03 Dec '12

Hello everybody, first of all, sorry for my english. I have a test server with the mod_cdr_mongodb backend configured. I'm trying to parse the cdrs with the mongodb driver for java, but i have a trouble with a part of the json that freeswitch sends (mongo saves the json document without errors). I haven't an API function to extract some data. Firstly I thought that the problem was in the java driver but I'm not sure. The problem is in the "callflow" part, with the json standard[1], a json document can't have two keys with the same name, but mod_cdr_mongodb builds the json with a 'n' callflow objects. Also it happends inside callflow object, with origination, originator and originatee objects. mod_cdr_mongodb saves the json with this format: http://pastebin.com/VRz6s0eb I modified the source of the backend adding json arrays around this objects and now I can parse without problems. The output with the modifications: http://pastebin.com/v05FN9Ta I'm in the correct way or I'm missing something? [1] http://www.ietf.org/rfc/rfc4627.txt Section 2.2 Regards -- Jose Fco. Irles Durá

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Freeswitch stable version 1.2.51
by baskar 03 Dec '12

03 Dec '12

Hi, In installed new version 1.2.51 but in that Readme it is mentioned that "This configuration, generally known as the "default configuration" for FreeSWITCH, is *NOT* designed to be put into a production environment without some important modifications. Please keep in mind that the default configuration is designed to demonstrate what FreeSWITCH *can* do, not what it *should* do in your specific scenario". To put into a production enviroment what are changes to be done. Could you please guide me to configure into a prodcution enviroment. I am using 1.0.6 stable version. Thanks in advance, N.Baskar -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-stable-version-1-2… Sent from the freeswitch-users mailing list archive at Nabble.com.

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FreeSWITCH Weekly News and Notes
by Michael Collins 03 Dec '12

03 Dec '12

Happy December to everyone! Last week was painful for many of us as we were dealing with a sustained DDoS attack on most of our infrastructure. Kudos to the guys for working through it. It seems the worst is over and we can get back to the business at hand: doing FreeSWITCH stuff. :) In spite of the drama last week we did have a conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_28>and we released 1.2.5.2! We discussed mostly the details of the DDoS we experienced and how the community can assist in the future so that we can mitigate the effects of such an occurrence. With the community's help we will be more resistant to the effects of any future attacks. We appreciate the outpouring of support we received from everyone. This week <http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_05> we will go back to discussing FreeSWITCH. We are still finalizing future guests so this week we'll do another installment of tips and tricks from the FreeSWITCH community. Among other things I will be showing how Chris Rienzo (IRC: crienzo) and I used the source this weekend to figure out what the XML preprocessor can do and get the wiki updated. I'll then show a simple example of the always-present-but-previously- undocumented command can do. As an added bonus we'll have an update on the ClueCon <http://www.cluecon.com> 2012 videos! Thanks and have a great week. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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XMPP Presence?
by Gavin Henry 03 Dec '12

03 Dec '12

Hi all, We're running our own XMPP chat server and would like to tie this in to FreeSWITCH. Has anyone tested presence via XMPP? The wiki is very light on this and would be happy to update. Thanks.

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