- FreeSWITCH-dev - FreeSWITCH Mailing Lists

mod_shout problem
by Babak Yakhchali 31 Aug '12

31 Aug '12

Hi I was trying to use telecast but it seems the table and urls are not produced correctly. After I read the mod_shout code I saw that in line 1414 a "select * from channels" query is used and in it's corresponding callback indexes are not correct so wrong values are used in creation of urls and the table. I created an issue on jira (But I donno if I did it correctly so I posted here) thanx

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Re: [Freeswitch-dev] [Freeswitch-users] Doubts about T38 passthru
by François Delawarde 30 Aug '12

30 Aug '12

Hi, Sent to the wrong ML, this probably is more a dev question: On Wed, 2012-08-29 at 10:46 +0200, François Delawarde wrote: > Hi, > > Whenever my provider puts me on hold, it reINVITES with an SDP > containing both audio (sendonly) and image media like so: > > ... > m=audio 5320 RTP/AVP 18 3 8 0 96 > c=IN IP4 <ip_address> > a=rtpmap:18 G729/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=sendonly > a=rtpmap:96 telephone-event/8000 > m=image 5322 udptl t38 > c=IN IP4 <ip_address> > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxRateManagement:transferredTCF > > > Not sure if it's a non-standard SDP, but that's what I get. Now when > t38_passthru=true, it only forwards the "image" part to the other leg. > When t38_passthru=false, it only forwards the "audio" part. > > Is there a way to make t38_passthru forward the audio part as well? Or > should I just change to a better provider? Thanks, François.

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added mp3 codec, please review
by Seven Du 30 Aug '12

30 Aug '12

Hi, To record mp4 with libmp4v2, I found the standard audio format is AAC but FS does't support, so I tried added a mp3 encoder, hopefully it works. I have successfully record into mp4 but audio doesn't playback, videos looks fine. Anyone can help review the code and give some input is appreciated. http://fisheye.freeswitch.org/changelog/freeswitch.git?cs=084dac2540a15bfc0… Playback Errors in VLC looks like this: peg_audio decoder debug: emulated startcode (no startcode on following frame) [0x10289faf0] mpeg_audio decoder debug: emulated startcode [0x10289faf0] mpeg_audio decoder debug: emulated startcode [0x10289faf0] mpeg_audio decoder debug: emulated startcode (no startcode on following frame) [0x10289faf0] mpeg_audio decoder debug: emulated startcode [0x10289faf0] mpeg_audio decoder debug: emulated startcode [0x10289faf0] mpeg_audio decoder debug: emulated startcode (no startcode on following frame) [0x10289faf0] mpeg_audio decoder debug: emulated startcode [0x10289faf0] mpeg_audio decoder debug: emulated startcode (no startcode on following frame) [0x105f10540] main audio output warning: buffer way too early (-126508), clearing queue [0x105f10540] main audio output warning: timing screwed, stopping resampling [0x10289faf0] mpeg_audio decoder debug: emulated startcode (no startcode on following frame) [0x10289faf0] mpeg_audio decoder debug: emulated startcode [0x10289faf0] mpeg_audio decoder debug: emulated startcode [0x10289faf0] mpeg_audio decoder debug: emulated startcode [0x10289faf0] mpeg_audio decoder debug: emulated startcode (no startcode on following frame) I use mod_mp4v2(C) with libmp4v2 to record mp4 file, if also been interested I could share on github or check into the video-media-bug branch. But I really want to merge with mod_mp4 though which is C++ based. @Paulo, I also CC you this message as I haven't figure out how to write hint track you input is welcome. Thanks. -- Seven Du Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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FreeSWITCH Conference Call Today
by Michael Collins 29 Aug '12

29 Aug '12

Hello all, We are having a nice discussion today: http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22 Mitch Capper and others who have experience will be talking to us about TLS, certs, CA's and things like that. Hopefully we'll all be better prepared to enable TLS transport on our devices and help those who come to the IRC and ML with TLS questions. 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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FreeSWITCH Weekly News and Notes
by Michael Collins 28 Aug '12

28 Aug '12

Hello all! Apologies for the delay on this week's news and notes. Yesterday was quite busy, but things are going well. I am happy to report that we have submitted the first chapter of the new FreeSWITCH book to the publisher! We've also conferred with a few members of the community and convinced Packt<http://www.packtpub.com>to let us add some bonus content! Stay tuned for more previews and teases. Right now it's still early in the game so I don't want to reveal too much. On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22>we discussed a number of things. First, we did a follow up to Ken's previous discussion <http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_15>about the stable 1.2 branch and using git. Second, we talked a bit about Vestec <http://www.vestec.com> and the great ASR application contest. (More information is forthcoming!) Lastly, we had Mitch Capper discussing the latest version of the FSClient <http://wiki.freeswitch.org/wiki/FSClient>Windows softphone. If you haven't tried it out I highly recommend it. It's now stable and feature-enabled to the point that I've discontinued using X-Lite or Jitsi. Tomorrow we hope to have a discussion about TLS<http://wiki.freeswitch.org/wiki/Tls>. We have several community members who are experienced with key and certificate management and we will be calling upon them to share their experience with the rest of us. After that we will have an open discussion. Cheers! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Re: [Freeswitch-dev] How to write an Application to control freeSWITCH
by Ali R. 28 Aug '12

28 Aug '12

http://www.amazon.com/FreeSWITCH-1-0-6-Anthony-Minessale/dp/1847199968/ref=… http://www.amazon.com/FreeSWITCH-Cookbook-Anthony-Minessale/dp/1849515409/r… ________________________________ From: Subhash <gvvsubhashkumar(a)gmail.com> To: aroumie(a)yahoo.com Sent: Monday, August 27, 2012 8:16 PM Subject: Re: [Freeswitch-dev] How to write an Application to control freeSWITCH Hi, Thanks for the reply. Can you please name the books and where can i get them. Thanks, Subhash. On Tue, Aug 28, 2012 at 2:59 AM, Ali R. <aroumie(a)yahoo.com> wrote: I can only speak for my experience with FS. As you mention FreeSwitch is a platform that means not a basic app. I recommend starting by reading the 2 books about FreeSwitch if you don't like to go through the Wiki jungle. Once you understand the main modules and the flow of events you can move on to the next phase where you can either write a module in C++ to control it from within the same process or use the powerful event socket in inbound mode for total control from anywhere else... > > > From: Subhash <gvvsubhashkumar(a)gmail.com> >To: Subhash kumar <gvvsubhashkumar(a)gmail.com> >Sent: Monday, August 27, 2012 8:19 AM >Subject: [Freeswitch-dev] How to write an Application to control freeSWITCH > > > > >Hi All, > > > I am new to freeSWITCH platform. > > > I want to write an application using c++ to control freeSWITCH so please guide me. > > Thanks in advance. > >Thanks, >Subhash. > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com/ > > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com/ > >Official FreeSWITCH Sites >http://www.freeswitch.org/ >http://wiki.freeswitch.org/ >http://www.cluecon.com/ > >FreeSWITCH-dev mailing list >FreeSWITCH-dev(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >http://www.freeswitch.org/ > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-dev mailing list >FreeSWITCH-dev(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >http://www.freeswitch.org/ > >

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How to write an Application to control freeSWITCH
by Subhash 27 Aug '12

27 Aug '12

Hi All, I am new to freeSWITCH platform. I want to write an application using c++ to control freeSWITCH so please guide me. Thanks in advance. Thanks, Subhash.

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libfreeswitch as embedded IVR
by Saju Pillai 27 Aug '12

27 Aug '12

Hello, I have to implement an IVR embedded in an existing app. My understanding is that I can declare global handlers for channel states to *intercept* calls and use the switch_ivr_* to do my processing. I have to either use tts or play recorded prompts to the caller and receive dtmf input from the caller. There can be multiple to & fro interactions with the caller in 1 call, this is determined by other data from the rest of the app. 1) Which channel state(s) should I hook into? 2) Will I have to setup a dialplan to make this work ? 3) Is there a sample/test or psuedo code that I can look at ? regards srp

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Re: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 74, Issue 12
by Craig N. Zimmerman 24 Aug '12

24 Aug '12

Hi Steve We have successfully done this with 900+ callers in a little different setup. Here is how we do it: The call comes into the first fs box then at about 300-350 callers. I have a script that sends a sip call to another box so we have an audio link then we forward all additional incoming calls to that box till with bypass media till it is maxed out then to the next box and then the next. You should be able to get up to 400 callers on one box depending on your hardware, bandwidth and codec. Also it helps to rent dedicated boxed to deal with the timing issues. Also we don't do the web interface but you can do that with a mysql db to keep track of which box the call is on. The only reason we have to this is because we can't get enough bandwidth into a single box. The other thing that I would like to do some time with this setup is hook my scripts up to rackspace or amazon's api and spin up more servers after I run out of physical boxes during peak load times. Thanks Craig I hope this helps. -----Original Message----- From: freeswitch-dev-bounces(a)lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of freeswitch-dev-request(a)lists.freeswitch.org Sent: Friday, August 24, 2012 12:00 PM To: freeswitch-dev(a)lists.freeswitch.org Subject: FreeSWITCH-dev Digest, Vol 74, Issue 12 hummm, interesting, i was just wondering if it is possible to make a tree shape conferencing bridge which would work something like this. 1. We will have many freeswitch boxes, each will running a single conference of its own up to say 50 users. These freeswitch box may be geographically distributed as needed, (though it may cause significant latency). 2. Then there is a freeswitch box running on top, all freeswitch boxes dial this top level freeswitch box as normal call. On top level freeswitch these calls will be answered and joined in a single so called top level conference. The second leg of this call will be joined to conference already running in lower level freeswitch boxes. 3. The main speaker / teacher of this virtual class room will just dial to top level freeswitch and join in the top level conference.. 4. For listeners / students of this virtual class to dial in and join, we can setup an OpenSIPs / Kamailio boxes which authenticates users first then forwards call to lower level freeswitch boxes. 5. All student calls will be mute when they join the conference. The teacher / speaker will have a web management interface which will query each lower freeswitch box and generate a participants list. Teacher will be able to unmute any student to listen to their questions etc. and then mute them again to answer the question etc. What do you guys think? Please suggest. Thank you. On Fri, Aug 24, 2012 at 1:15 PM, Stephen Dame <sdame(a)207me.com> wrote: I know a few commercial companies have scaled freeswitch to handle 250-500 callers in a single conference across boxes with direct audio cable connections in data center. I have an existing application that uses speex16(flash voip) and I can get 50 callers in a single conference before cpu gets to 70% on a c1.medium instance. I know I can throw more hardware at the problem, but interested in bridging the same conference number between multiple freeswitch instances and presenting single ESL notifications back to the existing application so all users events are seen. This is a distance learning app, and not business audio conference, so some latency is tolerable. I would locate the multiple freeswitch servers in same zone and possibly use the new high I/O EBS instances running in SSD. Any thoughts on this? Regards, Stephen _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-dev mailing list FreeSWITCH-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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Scaled or distrbuted mod_conference?
by Stephen Dame 24 Aug '12

24 Aug '12

I know a few commercial companies have scaled freeswitch to handle 250-500 callers in a single conference across boxes with direct audio cable connections in data center. I have an existing application that uses speex16(flash voip) and I can get 50 callers in a single conference before cpu gets to 70% on a c1.medium instance. I know I can throw more hardware at the problem, but interested in bridging the same conference number between multiple freeswitch instances and presenting single ESL notifications back to the existing application so all users events are seen. This is a distance learning app, and not business audio conference, so some latency is tolerable. I would locate the multiple freeswitch servers in same zone and possibly use the new high I/O EBS instances running in SSD. Any thoughts on this? Regards, Stephen

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