September 2006 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch now has Python Support
by Brian Fertig 20 Oct '06

20 Oct '06

The framework is there now for Python. It still has some hiccups but I think we can have them cleared up this weekend. Have questions let us know! Brian Fertig

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FreeSwitch.Net
by Martin Sørensen 30 Sep '06

30 Sep '06

Hi List... can somebody help me get started with Freeswitch.Net step by step... Configuration of FreeSwitch compiling and a sample module I c# Regards Martin

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Re: [Freeswitch-users] decent VOIP provider
by Charleseds＠yahoo.com 27 Sep '06

27 Sep '06

call this number 5164256408 Sadiq <sadiq(a)voicesky.com> wrote: Hi, You can Try VOICE.SKY Communications u can email us on noc(a)voicesky.com Sadiq Sr.Network Engineer VOICE.SKY - NOC,R&D Sadiq(a)voicesky.com -----Original Message----- From: Sebastian Replanski [mailto:sreplans@hotmail.com] Sent: Monday, September 25, 2006 11:27 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] decent VOIP provider Hi List, I am looking for recomendations on a good VOIP provider to terminate my FS calls to the PSTN in the US. I need a company that can handle hundreds of calls originating at once. so far I have tried: binfone, iaxtel, sixtel, vitelity, voipjet and bandwith.com the last one is the only one that works, but its very expensive (of course). any ideas? thanks _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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STUN server implementations
by Peter Saint-Andre 26 Sep '06

26 Sep '06

As you may know, we in the Jabber community are working on Jingle, a way to do VoIP signalling over XMPP. As part of this effort we're using the ICE methodology defined by the IETF, so we want to start running a STUN server at jabber.org. Unfortunately, I'm having a hard time finding a reliable, well-maintained, open-source STUN server implementation. So I'm wondering: 1. Can anyone on this list recommend a STUN server implementation? 2. Will FreeSWITCH eventually include a STUN server implementation? Thanks! Peter -- Peter Saint-Andre Jabber Software Foundation http://www.jabber.org/people/stpeter.shtml

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Re: [Freeswitch-users] decent VOIP provider
by Michael Huynh 26 Sep '06

26 Sep '06

I've tried voxee.com. Their rates are excellent. Unfortunately, their service varies from good to sometimes not working. That's an unacceptable level of service for us, however, this might work for you. We're currently using gafachi.com. Although expensive, it is 100% reliable with great voice quality. good luck >From: "Sadiq" <sadiq(a)voicesky.com> >Reply-To: freeswitch-users(a)lists.freeswitch.org >To: <freeswitch-users(a)lists.freeswitch.org> >Subject: Re: [Freeswitch-users] decent VOIP provider >Date: Tue, 26 Sep 2006 09:43:21 +0300 > > >Hi, > > >You can Try VOICE.SKY Communications u can email us on noc(a)voicesky.com > > > >Sadiq >Sr.Network Engineer >VOICE.SKY - NOC,R&D >Sadiq(a)voicesky.com >-----Original Message----- >From: Sebastian Replanski [mailto:sreplans@hotmail.com] >Sent: Monday, September 25, 2006 11:27 PM >To: freeswitch-users(a)lists.freeswitch.org >Subject: [Freeswitch-users] decent VOIP provider > >Hi List, > >I am looking for recomendations on a good VOIP provider to terminate my FS >calls to the PSTN in the US. I need a company that can handle hundreds of >calls originating at once. > >so far I have tried: >binfone, iaxtel, sixtel, vitelity, voipjet and bandwith.com >the last one is the only one that works, but its very expensive (of >course). > >any ideas? > >thanks > > > >_______________________________________________ >Freeswitch-users mailing list >Freeswitch-users(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > >_______________________________________________ >Freeswitch-users mailing list >Freeswitch-users(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org

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decent VOIP provider
by Sebastian Replanski 26 Sep '06

26 Sep '06

Hi List, I am looking for recomendations on a good VOIP provider to terminate my FS calls to the PSTN in the US. I need a company that can handle hundreds of calls originating at once. so far I have tried: binfone, iaxtel, sixtel, vitelity, voipjet and bandwith.com the last one is the only one that works, but its very expensive (of course). any ideas? thanks

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這裡有使用中文的朋友嗎?
by will 22 Sep '06

22 Sep '06

這裡有使用中文的朋友嗎? 先跟大家問好一下，因為這是很難得的事情！ Will Lien

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Win32 ver
by Duane Whittingham 22 Sep '06

22 Sep '06

Hi, a while ago someone compiled and posted a Win32 version, I was wondering if theres a newer one and if so a d/l link please, tnx. Duane

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Re: [Freeswitch-users] How many concurrent calls
by karl 22 Sep '06

22 Sep '06

Is it stablee .. as of yetr -----Original message----- From: "Muhammad Shahzad" shaheryarkh(a)googlemail.com Date: Sat, 23 Sep 2006 01:20:06 +1000 To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] How many concurrent calls > Hi Everyone, > > Sorry i was out-station and extremely busy for last couple of weeks or so, > therefore, i couldn't reply earlier. Anyways, here is some data regarding > our testing of FreeSWITCH conducted at a NOC in Lahore, Pakistan, > > Setup: > 1. We setup two servers of same hardware i.e. *Fujitsu Siemens Xeon *3.2 GHz, > each running Gentoo Linux 2006.0 with kernels 2.6.15-r8 and > 2.6.17-r5respectively. > 2. These servers were then connected over cross-line cable to each other > using Gigabit NICs. > 3. We installed Asterisk version 1.2.10 on machine 'A' and FreeSWITCH > snapshot dated 2006-09-01 on machine 'B'. > 4. Each machine had another Gigabit NIC through which it was connect to > Internet, so that i can access them remotely. > 5. I was connected to these servers over 100kbs CDMA wireless Internet link > via ssh. > 6. Date was Sept. 02, 2006, 0300hours PKST (GMT+5 zone). > > Test Plan I: (Simple Playback) > We generated batches of 100, 200, 300, 400 and 500 calls, with steps 25, 50, > 100 calls dialed simultaneously at an interval of 10 seconds. Each call > plays a flat tone of one minute length, file format WAV, codec GSM. > > Maximum concurrent calls should be, > > X | 25 50 100 (steps) > ------------------------------------------ > 100 | 100 100 100 > 200 | 150 200 200 > 300 | 150 250 300 > 400 | 150 250 400 > 500 | 150 250 500 > (Test Matrix: Fixed interval of 10 seconds) > > In this plan number of calls equal number of active IAX2 channels. Maximum > time for maxium concurrent calls was between 9 and 10 seconds. > > Results: > All batches successfully executed, however at 400 or above concurrent calls > the load averge shoot to over 5.6 and FreeSWITCH console was showing call > details so fast that my remote ssh link hanged. When the link resumed the > FreeSWITCH was still up and number of calls were decreasing as the test was > moving towards end. > > Test Plan II: (Forward Bridged Calls) > 1. We generated same batches of calls except that now they dials out to > Machine A (the one running Asterisk). That machine accepts the call and > plays back a milliwatt for 1 minute, codec GSM. > > 2. Since calls were bridged so number of calls were half the number of > channels. > > Results: > At 436 active channels (218 concurrent calls) Asterisk crashes and therefore > all calls dropped. > > Test Plan III: (Reverse Bridged Calls) > 1. Same as plan II, except now we generated calls from Machine A (i.e. > Asterisk) and Machine B (i.e. FreeSWITCH) answers the call and plays back > flat tone of 1 minute length (file format WAV, codec GSM) > > 2. Since calls were bridged so number of calls were half the number of > channels. > > Results: > At 394 active channels (197 concurrent calls) FreeSWITCH crashes and > therefore all calls dropped. > > Warning: > These tests may not exhibit same results in all environments, we did not > follow any specific testing standard nor we endorse any software or hardware > used in these tests. > > I wanted to decrease the interval as well as call steps to get more exact > information on how and when FreeSWITCH and asterisk crashes but i had to > abandon further plans due to bad weather (i was using wireless link which > went down). > > Now i am planing for FreeSWITCH to FreeSWITCH bridged calls testing, i will > send you the results whenever i got those servers available to me. > > Thanks and best regards. > > -shahzad > > > On 9/4/06, Brian West <brian.west(a)mac.com> wrote: > > > > Was this 500 bridged calls? or 250 bridged calls? or 500 going to > > playback? > > /b > > > > On Sep 3, 2006, at 5:05 PM, Muhammad Shahzad wrote: > > > > Not sure, but i think that depends on your hardware. Freeswitch has no > > such built-in limitation on number of concurrent calls. > > > > We have tested it upto 500 concurrent IAX2 calls on an Intel Xeon 3.2 GHz > > machine with 2.0GB of RAM. :-) > > > > -shahzad. > > > > > > On 9/2/06, Kamran Ahmad <kamranah(a)gmail.com> wrote: > > > > > > HI > > > > > > How many concurrent calls are supported by freeswitch on linux or > > > windows ? any idea? > > > > > > Thanks > > > Kamran > > > > > > > > > > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users(a)lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > -- > > Muhammad Shahzad > > ------------------------------- > > Manager Support, > > Convergence Business Systems, > > Lahore, Pakistan. > > Cell: 0092 334 422 40 88 > > Email: shaheryarkh(a)googlemail.com > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > -- > Muhammad Shahzad > ------------------------------- > Manager Support, > Convergence Business Systems, > Lahore, Pakistan. > Cell: 0092 334 422 40 88 > Email: shaheryarkh(a)googlemail.com >

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Is Freeswitch better than Asterisk for me?
by Barry Roberts 21 Sep '06

21 Sep '06

I've been using Asterisk as simple outbound caller for a small (<50) group of people with very simple IVR (press 1 to ack, press 2 to replay). It works ok, but I haven't come up with a good way to keep both my outbound IAX trunks busy to get through the calls as quickly as possible. I'm copying .call files into Asterisk's outgoing directory, and I have a script that makes sure there are 2 files there, but as soon as I get a busy/no-answer, there's a line that is under-utilized. Does freeswitch have better way to do this? I spent several years as an IVR developer using Natural Microsystems and Dialogic cards. So I'm quite accustomed to that level of programming, as opposed to the PBX-like way Asterisk works. I would love to find a way to do that low-level coding for my IAX trunks. TIA, Barry Roberts

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