August 2012 - freeswitch-dev@lists.freeswitch.org

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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[Freeswitch-dev] 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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[Freeswitch-dev] FreeSwitch / M3UA
by Paulo Vicentini 23 Aug '12

23 Aug '12

Is there any plan to support M3UA in FS? Media Gateways supports M3UA/H.248 (e.g: Audiocodes) and it would be fine to interoperate directly with them with FS: FS (SGC)<---SIGTRAN (M3UA)/Megaco ---> AudioCodes SGW (MTP1,2)<----link set ss7 FS would need a Sigtran/M3UA module to communicate with signaling gateway (AudioCodes M8k) and a Megaco(H.248) module to control audio MGW (Notice that MTP1,2 isn't handled in FS - M3UA(ISUP) ) Any comments on that? Thanks Paulo

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Re: [Freeswitch-dev] Channel_Bridge event
by Peter Olsson 23 Aug '12

23 Aug '12

I believe the event only arrives when they are actually talking to each other, and that is when both parties are answered. However, it's totally valid to do uuid_bridge if at least one call is answered. /Peter Från: freeswitch-dev-bounces(a)lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] För Papineni, Suneel Skickat: den 23 augusti 2012 16:56 Till: freeswitch-dev(a)lists.freeswitch.org Ämne: [Freeswitch-dev] Channel_Bridge event Hi, I am expecting a "Channel_Bridge" event when I issued "uuid_bridge" command between two call legs (one call is active and answered and other call is in ringing state), but I am not getting this event. I tried using fs_cli.exe to view the events. Only when other party answered, then I could see Channel_Bridge event. Can someone help me in understanding why I am not getting this event when Bridge command is issued? As per uuid_bridge command wiki, call should get bridged if at least one party is answered and active. Thanks& Regards Suneel !DSPAM:503641c532762417717798!

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[Freeswitch-dev] Channel_Bridge event
by Papineni, Suneel 23 Aug '12

23 Aug '12

Hi, I am expecting a "Channel_Bridge" event when I issued "uuid_bridge" command between two call legs (one call is active and answered and other call is in ringing state), but I am not getting this event. I tried using fs_cli.exe to view the events. Only when other party answered, then I could see Channel_Bridge event. Can someone help me in understanding why I am not getting this event when Bridge command is issued? As per uuid_bridge command wiki, call should get bridged if at least one party is answered and active. Thanks& Regards Suneel

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[Freeswitch-dev] Creating a session without incoming or outgoing call
by saurabh nirkhey 22 Aug '12

22 Aug '12

I am using mod_skel and writing my own module. My application needs me to be able to create a session without an incoming call. My questions are: 1. Is it possible to do this? 2. Do I need to inherit CoreSession or does an easier alternative exist? 3. Can I invoke my application from eventsocket without originating a call at all? Thanks for any inputs. Regards, Saurabh Nirkhey

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[Freeswitch-dev] FreeSWITCH Community Conference Call Today (Important Information - Please Read)
by Michael Collins 22 Aug '12

22 Aug '12

Hello all! The conference will start in less than an hour. Here's our agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22 We've made a change to the system. Everyone will come in muted. When you press 0 it will send a message into the #FreeSWITCH IRC channel - "user has a question". One of the conference moderators will unmute you so that you can ask your question. We are hoping this will make the conference more enjoyable for everyone. 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-dev] SLA lights not working on some inbound calls
by Don Dawson 22 Aug '12

22 Aug '12

I have 3 Cisco SPA504G phones connected to Freeswitch using the SLA feature. Everything seems to work except for inbound calls, some calls the lights are proper and others the lights are wrong. I have 3 phones sharing extension 122. The call sequence to demonstrate the behavior of the lights, from a fourth phone, I dial 122 and all 3 phones ring with the x122 line key blinking. If I answer the phone that is first in the list of INVITEs to x122, then the other 2 phones work properly with their x122 line key blinking indicating an answered call. However if I answer on the phone that was the 2^nd or 3^rd INVITE, then the line key lights for the other 2 phones go out indicating an idle line (it should show an active line). Looking at the SIP trace, for the call that worked properly, there are 3 INVITEs to the phone, 200 OK from the phone that answered (which is the phone that was the target of the first INVITE), CANCEL to the other 2 phones, then the key part, the NOTIFYs are sent with appearance-state=active. When I answer with one of the other 2 phones, the SIP trace shows that same sequence EXCEPT the NOTIFYs have appearance-state=idle. I tried this with the latest get, version 1.2.1 and a get from about 9 months ago. All have the same issue.All phones register using domain name. Here is a portion of the SLA debug showing 'idle' that when works the call_info_state is 'active': 2012-08-21 14:52:13.081302 [ERR] sofia_presence.c:1249 STATE SQL update sip_dialogs set call_info='appearance-index=1',call_info_state='idle' where hostname=....................... What should I try?

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[Freeswitch-dev] FreeSWITCH Project Looking For Volunteers: Web Site Dev/Maint
by Michael Collins 21 Aug '12

21 Aug '12

Hello all, I need to refresh my list of people who have Web dev skills and are willing to assist with things like Web site design, programming, and maintenance. If you have any of the following skills, or know someone who does and who is willing to donate a few hours, please email me offlist: HTML5 CSS2/3 Javascript/JQuery Drupal/PHP Wordpress/PHP Django/Python Web design/graphics Apache administration in LAMP environment Please note that not every single skill is currently in demand; we are trying to get a feel for who knows what and hopefully anticipate future needs. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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[Freeswitch-dev] FreeSWITCH Weekly News and Notes
by Michael Collins 20 Aug '12

20 Aug '12

FreeSWITCH Weekly News and Notes is back after a brief hiatus. In case you hadn't heard: FreeSWITCH 1.2 is out!<http://freeswitch.org/node/410> In fact, Anthony and Ken are working on a 1.2.2 release. Stay tuned for an announcement. On last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_15> we discussed some of the git commands you may need to run in order to get yourself moved up to the 1.2stable branch<http://fisheye.freeswitch.org/changelog/~br=v1.2.stable/freeswitch.git>. Join us this Wednesday<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22> and we'll do a quick follow up for those who may still have questions. We hope to have some other announcements as well. In post-ClueCon news we'd like to let everyone know that Vestec<http://vestec.com/> is finalizing the arrangements for the great ASR (automated speech recognition) app-building contest. This is a great opportunity to get some cash and free speech recognition licenses in return for investing some time and effort into learning the Vestec system and building an application to show off to the world. It's also a great way to help promote FreeSWITCH among larger enterprises who may not realize that professional-grade ASR is available. We will discuss this further on Wednesday's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22> . Lastly we'd like to let everyone know that the ClueCon videos will be made available in the coming weeks and months. Please give us some time to do a little editing before we release them all. It will be worth the wait! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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