August 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Video Support (Samuel Macedo)
by Jeff Dworkin 09 Aug '10

09 Aug '10

The DiaStar Server is all software, just download and install from the iso. You can do single port video with just what is on the iso. If you want to test conferencing you have to fill out the form at http://www.dialogic.com/technologies/open-source/diastar.htm?regID=38692 -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request(a)lists.freeswitch.org Sent: Monday, August 09, 2010 4:05 PM To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 50, Issue 45 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: GPL Wins Again (Madovsky) 2. Asterisk to Freeswitch (Michael Gende) 3. Video Support (Samuel Macedo) 4. Re: Modems (Ken Gillett) 5. Re: Custom Headers (Jesse Cloutier) 6. Re: Video Support (John Hermanski) 7. Re: Video Support (Samuel Macedo) ---------------------------------------------------------------------- Message: 1 Date: Mon, 9 Aug 2010 12:59:58 -0400 From: "Madovsky" <infos(a)madovsky.org> Subject: Re: [Freeswitch-users] GPL Wins Again To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Message-ID: <2F9511642CCD492CA057DB89F09BF420@MOBILEE1705> Content-Type: text/plain; format=flowed; charset="iso-8859-1"; reply-type=original its' also an excellent way for big companies to avoid to pay employees to make all the dev work also.... ----- Original Message ----- From: "M. Ranganathan" <mranga(a)gmail.com> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Monday, August 09, 2010 12:24 PM Subject: Re: [Freeswitch-users] GPL Wins Again > On Mon, Aug 9, 2010 at 12:08 PM, Madovsky <infos(a)madovsky.org> wrote: >> I read well don't worry, >> I say that because companies start to instrumentalize open source. >> so after years some developers (for google summer source code for ex) >> realise that they spent all their lifetime for a big company for free... >> that is why I called Open Fools.... >> >> >> ----- Original Message ----- > > I speak for myself (as an individual) here : > > I disagree with your characterization. Open source is engineers > collaborating informally on a shared piece of basic infrastructure > code so as to amortize engineering cost, testing cost and beta > testing. As an open source developer, you give away something to get a > lot in return so you can build on the sum total of collaborative > effort. > > Open source is also an excellent way for students to establish a track > record. Yes it is free labor but try getting a job these days as a > fresh graduate with no experience and you will see what I am talking > about. > > Ranga > > > -- > M. Ranganathan > > _______________________________________________ > 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 ------------------------------ Message: 2 Date: Mon, 9 Aug 2010 13:16:52 -0500 From: Michael Gende <mgende(a)gendesign.com> Subject: [Freeswitch-users] Asterisk to Freeswitch To: freeswitch-users(a)lists.freeswitch.org Message-ID: <AANLkTinmPjiFhkmxpNBVrmExgyScj=Wai=NA+Dds7E-q(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hello, I was wondering if someone could please give me a shove in the right direction concerning some "translations". I've got some dial plan examples for a vendor's handset that are created for Asterisk. Naturally, we use FreeSwitch. Any "dial plan translation" pages to be found? I found the Rosetta Stone on the FS site. Very comprehensive, but I didn't see things for the commands in my examples (i.e, Asterisk's Pickup = FS command X) I'll figure it out but wondered if someone else had already been down this road. Regards, Mike G. P.S. Nice new FS book, by the way. Got our copy last week, very instructive and a good read to boot.

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Custom Headers
by Jesse Cloutier 09 Aug '10

09 Aug '10

Hi all, Is there a way to read custom sip headers (eg: P-MyHeader or X-MyHeader) in the freeswitch dialplan? I am using freeswitch as a SBC and would like to pass certain variables to it for the cdr. Thanks! Jesse Cloutier

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FS performance
by Jean-Yves F. Barbier 09 Aug '10

09 Aug '10

Hi list, How does FS performs compared to Kamailio on the same machine? (I'm just talking about pure switching, no transcoding, nor AA) JY --

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Re: [Freeswitch-users] we are under attack
by Hristo Benev 09 Aug '10

09 Aug '10

If it is just one IP it is not that bad... First block it in iptables, then file a complain with your ISP(they will help you block at entry point) and his ISP. Fail2 ban works well with brute force...(or at least forces them to extent the attack) >-------- Оригинално писмо -------- >От: Seven Du >Относно: [Freeswitch-users] we are under attack >До: freeswitch-users >Изпратено на: Понеделник, 2010, Август 9 04:24:20 EEST >Hi, > >We suffered an SIP attack from 67.23.236.75. It attempted to register >to our SIP server using bruce force. > >We are running FS on a PC as our office PBX. When all phone failed, >we noticed a high CPU load with 90%+ waiting or nice, and in the >meantime it used up memory and start swapping to disk. > >It's a cheap PC with only 700MB memory, and we are running FS, DB, >Rails and other system on it. So it took me some time to check every >part. And it didn't help even I did a full server reboot. Finally I >turned on sip trace in FS and found thousands and millions of illegal >registers. And then I blocked the IP in iptables. > >During the hard time, I noticed: > >1) It stucks on one CPU even I have 2 core since sofia-sip is single threaded ? > >2) CPU also waiting page swap when used up memory. > >3) After I dropped all packets from that IP, FS still kept sending >register error sip messages for quite a long time before I restarted >FS. > >Now looking to add http://wiki.freeswitch.org/wiki/Fail2ban, hope this helps . > >Hope this helps if some one also suffered this. > >7. > >-- >Blog: http://www.dujinfang.com >Proj: http://www.freeswitch.org.cn > >_______________________________________________ >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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FreeSwitch - Skype call route
by Shamun toha md 09 Aug '10

09 Aug '10

A receive calls, B receive calls, they are operator (mod_skypopen) - Caller party 1 calls to A - A receive the call talk for 2 minute - A decide he needs to send the call two B How can A tell caller party 1 (instant), please hold on, i am forwarding this call to B? Can i do this? Is it done via dial plane? Can you please just hints me? Thank you Best regards

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FS and kernel PREEMPT_RT in Fedora 10
by Madovsky 09 Aug '10

09 Aug '10

Just tried PREEMPT_RT patch for kernel 2.6.33.7 on Fedora 10 64bits set on 1000HZ. Everything works fine -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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FreeSWITCH - Skype capacity
by Shamun toha md 08 Aug '10

08 Aug '10

Before moving to production, i am just collecting few information's not to get surprise in future. Therefore, can anyone provide estimated suggestion, how many channels/node can make server 100% reserve? System plan: =========== - Single Quad Core Intel Xeon Processor uit de 5400-reeks met maximaal 3,16 GHz - 8GM RAM Question: ======= 1. Can i handle 100 or 300 simultaneous calls for Skype IN and Skype Out using FS? Thank you Best regards

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Distributing SIP registrations using ODBC
by Dan Lane 08 Aug '10

08 Aug '10

Hi Guys, I have a cluster of FS boxes all sharing SIP registrations using ODBC and under certain circumstances I can register Client-A on FS-A and Client-B on FS-B and make calls between Client-A and Client-B without issue as the relevant FS box knows how to get a call to the client from the shared registrations table. However when the callee client is behind NAT (or using a SIP client configured to only allow calls from a server it has registered with) the call gets dropped by the NAT router (or client) I can fix this easily with some LUA but before I do I wanted to check if there was an advisable or recognised FreeSWITCH way to work around this. Has anyone else solved this issue already? Regards, Dan

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new install on freebsd 8.1
by D Cubi 08 Aug '10

08 Aug '10

Forgot to add URL of instructions followed in original posting to list :( Followed these instructions. http://wiki.freeswitch.org/wiki/Installation_Guide#FreeBSD All seemed to have installed properly, but can't find bin folder : # pwd /usr/local/freeswitch # ls -l total 8 drwxr-xr-x 11 root wheel 512 Aug 2 11:24 conf drwxr-xr-x 2 root wheel 512 Aug 2 11:24 htdocs drwxr-xr-x 2 root wheel 512 Aug 2 12:44 lib drwxr-xr-x 4 root wheel 512 Aug 2 12:12 sounds Does anyone have any idea where I went wrong? Thnx Darren

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FreeSwitch - Skype
by Shamun toha md 08 Aug '10

08 Aug '10

Hello, I have few questions about FS and Skype, and very confuse. Please kindly can someone give me some advise? Assume A, B, C: ============= - A to use skype messenger or mobile/land line numbers to place a call (A is a flying person, who can be in CAR/Train or office or home or restaurant) - B will always receive calls in sip phone or skype messenger (B is a static person, he is in computer, his job is operator role for every call) - C is call receiver, by B only ( C is flying person, who can be in CAR/ Train or even office waiting in Skype messenger) - A is caller, B is Operator, who decide if C will be connected, and C is waiting for a call. - A is logged in to skype messenger or not - B is logged in to skype messenger or sip phone to FS server - C is logged in to skype messenger or not - FS is installed with Skype incoming numbers + skypeID's Question: ======= Can caller A make call from his skype messenger to FS skypeID? How B will know about this call? How C can be connected by B? How the flow happens in FS and Skype? In terms of Sipphone vs Skype Messenger? Thank you Best regards

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