November 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Inbound calls question
by regs＠kinetix.gr 04 Nov '08

04 Nov '08

Hi, How can my FS accept inbound SIP calls from other gateways without the need of a registration from their part? I only need to be able to accept inbound calls from specific gateway IPs. I tried creating my own profile and gateway but it fails : "Error Creating SIP UA for profile: myprofile" Can someone give me some first-step directions?

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Scaling Question
by Eric Liedtke 04 Nov '08

04 Nov '08

Greetings, I'm working on a server in an SBC setup. It has a copper 10/100 for admin and 2 GiGe connections for private/public peering(SIP signaling and media). I've been testing with some traffic and am finding I hit a ceiling around 800-900 active calls. The system CPU ends up being about 40% which seemed really high to me. Is this simply due to the amount of networking happening with the rtp in and out the different interfaces or am I operating something in a non optimized fasion ? It's an e1000 based card, I've tried messing around with the interrupt throttling but in the end the dynamic setting seemed to work best. Given that a chassis has done 3k simultaneous calls with about half the processing power I currently have in this server I must be running sub optimal. Any idea's of where I might look to correct this ? kernel tuning ? network tuning ? Other than seeing a lot of irq's from the network cards I"m not sure what else might be using up the system cpu. To squeeze out as much as possible I have the core databases housed in a tmpfs partiion, and I've also disabled the sql scoreboard (-nosql). Any other areas I should be exploring to increase the number of calls this chassis can process ? Thanks in advance for any ideas -Eric

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Re: [Freeswitch-users] Freeswitch as SBC ?
by Tristan 04 Nov '08

04 Nov '08

Hi ram, Part of the solution I built is a redundant openser in front of a redundant freeswitch install. FS acting like an SBC ( ie, all calls comes through, and are filtered based on xml_curl dialplan rules ). Asterisk was not the solution for me, as it's not as easy as FS to scale and manage in my HA environnement. I'm currently thinking of adding some kind of failover method in FS, to provide 99.9999% uptime even if one box fails. The better way I saw atm, was to send to a spare FS instance everything going on the production box, allowing it to take over the prod'box in case of failure ( ie taking the ip, opening rtp ports that were opened, and keeping tracks of SIP transactions and their status ). I believe this sort of features cannot be implemented easily in asterisk, but in FS it should not be too much difficult. Maybe adding a new channel state tracking changes on the master it's spare of... ( why not building also a master/master cluster ? ) .... If you need more, just ask, Regards, Tristan. ram a écrit : > Hi > > what kind of setup you have > > One Openser and sevaral Asterisk boxes ? > > ram > > On Fri, Oct 31, 2008 at 3:32 PM, Tristan <tristan(a)telemaque.fr > <mailto:tristan@telemaque.fr>> wrote: > > Hello Andreas, > > I'm using it in production as an SBC for 3 call-centers in > combination with openser. > If you need help setting this up, just ask questions :) > > Regards, > > Andreas Dimitriou a écrit : >> >> Hello, >> >> >> >> Has anyone used Freeswitch as SBC ? >> >> >> >> BR >> >> Andreas D >> >> ------------------------------------------------------------------------ >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users(a)lists.freeswitch.org >> <mailto:Freeswitch-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org <http://www.freeswitch.org/> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > <mailto:Freeswitch-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org <http://www.freeswitch.org/> > >

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Transcoding to GSM using SOX
by Brian Wood 04 Nov '08

04 Nov '08

I'm trying to transcode a .wav to a .gsm using SOX v14.0.1. What I get is a completely garbled when using playback in FreeSWITCH. I have verified that at least one other application (Quicktime Player) can in fact play the resulting .gsm file. I have tried the following: sox william.wav -t gsm william.gsm sox william.wav -t gsm -r 8000 william.gsm resample -ql sox william.wav -t gsm -r 8000 -c1 william.gsm resample -ql And I am playing the file using: originate sofia/internal/1000(a)192.168.1.1 &playback(/home/user/william.gsm) Other formats such as uLaw and aLaw (-t ul and -t al, respectively) seem to work fine. Does anyone know of a set of parameters for SOX that will make this work? At the end of the day, I just want it to work, so if anyone can recommend any other known/good command line tools to encode GSM that would be fine too.

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Re: [Freeswitch-users] VxML Parser?
by Douglas Garstang 04 Nov '08

04 Nov '08

Thanks. Maybe I need to go and get my hands more dirty, but how would freeswitch interface to this? Doug. ________________________________ From: EdPimentl <edpimentl(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, November 2, 2008 1:39:22 PM Subject: Re: [Freeswitch-users] VxML Parser? Here.. The most active VoiceXML projects... http://jvoicexml.sourceforge.net/ http://jvoicexml.sourceforge.net/documentation.htm http://www.commetrex.com/products/ctmiddleware/bladewarevxml.html -E

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RTMP Support (Flash)
by Saurabh Aggarwal 04 Nov '08

04 Nov '08

Any plans on adding RTMP support to freeswitch - that could be a real killer feature, allowing flash clients to call into Freeswitch. Now that Red5 has done all the hard work, it would be pretty cool if an endpoint can be developed. -Saurabh _________________________________________________________________ Want to read Hotmail messages in Outlook? The Wordsmiths show you how. http://windowslive.com/connect/post/wedowindowslive.spaces.live.com-Blog-cn…

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Re: [Freeswitch-users] get channel status
by B Karthik 04 Nov '08

04 Nov '08

Set hangup_after_bridge variable to false and run the info command after bridge, you will see a lot of variables with the channel's current status. You could later reference those variables directly in your xml / script dialplan. ---------- Forwarded message ---------- > From: "Gopal krishnan" <saigop(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Mon, 29 Sep 2008 15:40:59 +0530 > Subject: [Freeswitch-users] get channel status > Hi, > > Is there any possibilities that I can check my channel status whether it > is ringing or answer or hangup. I am trying to fetch thru uuid but couldn't > able to do that. > > any suggestion would be helpful. thanks > > -- > Thank you with regards, > Gopal, > >

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stun.freeswitch.org: No route to host
by Birgit Arkesteijn 04 Nov '08

04 Nov '08

Hi, Is stun.freeswitch.org down? Our FreeSWITCH installation broke overnight (no configuration changes made) because stun can no longer be reached. # telnet stun.freeswitch.org 3478 Trying 216.27.14.70... telnet: connect to address 216.27.14.70: No route to host # traceroute stun.freeswitch.org ends with .. 18 * dr1-g2-2.cry.hostedsolutions.com (216.27.30.86) 76.936 ms 77.497 ms 19 freeswitch01.bandwidth.com (216.27.14.70) 81.201 ms !X 81.730 ms !X 81.900 ms !X Cheers, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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Playing a sound file (or TTS) to the called party when transferring a call just before bridging
by Gonzalo Servat 03 Nov '08

03 Nov '08

Hi All, This is basically what I'm trying to do: - Caller dials in and FS runs the Lua script I'm writing - Main menu is played to the caller - Caller presses 1 - Still in Lua, i run session:transfer() to an extension of another dialplan context - When the called party picks up, a wav file (or TTS) is played to the CALLED party - Once the sound stops playing, the caller is bridged to the called party To achieve this, the nice folks at #freeswitch suggested a few things: 1) execute_on_answer. This doesn't appear to work ... no application seems to be executed when the called party picks up. Is it because the call is already answered when the caller originally dials in at the very beginning? 2) bridge_pre_execute_bleg_app / bridge_pre_execute_bleg_data. This is the closest I've come to getting it working. I used the following in the Lua script: if( selection == "1" ) then session:setAutoHangup(false) session:execute( "set", "bridge_pre_execute_bleg_app=speak" ); session:execute( "set", "bridge_pre_execute_bleg_data=flite|kal|this is kal saying something" ); session:transfer( "10", "XML", "extensions" ) end I actually see in the logs that it's "saying text: this is kal saying something" but it goes no further. On the called party side, I don't hear the text nor is the call bridged. I tried using "playback" instead of "speak" to see if it was a TTS problem but no, same result. Any ideas? Thanks Gonzalo

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Re: [Freeswitch-users] VxML Parser?
by Michael S Collins 03 Nov '08

03 Nov '08

Doug, Welcome to FreeSWITCH! BTW, I owe you a thank you because I first heard of FS from one of your posts to the asterisk users list. :) FS is definitely designed to be extensible. In fact, one night when Tony was ticked off about a post earlier in the day, he whipped up mod_yaml in three hours! However, I don't believe anyone has yet sung the virtues of VXML to the point of imploring the devs to add it, nor has anyone put up a bounty. My guess is that VXML would be added at some point. It will happen more quickly if Tony doesn't have to do it gratis if know what I mean. Hey are you still a Python guy? Just curious if you have played with mod_python or wikipbx yet. -MC Sent from my iPhone On Nov 2, 2008, at 1:15 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > Well, the same could be said about Asterisk and yet there are third > part vxml parsers (although not very good) available for it. > > I have developers that want to write voice applications directly in > vxml. They want to use vxml, not js, lua, python or anything else. > That's a hard requirement. > > I'm surprised freeswitch doesn't have this available. Being able to > rapidly deploy voice applications with vxml doesn't seem like an > unusual requirement. > > Doug. > > From: Andrew Gilbert <gilbertandrew(a)mac.com> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Sunday, November 2, 2008 12:51:31 PM > Subject: Re: [Freeswitch-users] VxML Parser? > > Quick answer, it's a switch/b2bua and not a vxml parser. > > Longer answer involves questions about what you are trying to do. > > If you need quick voice apps, check out the many options including > js, lua, python, liverpie, etc. It is a rich set of options and much > of this may be more appropriate / productive than full vxml > dependent on requirements. > > If you have a hard requirement / need for vxml might check out > voiceglue.org. This is an asterisk related project, but in theory > and with some effort who knows. > > > > > > On Nov 2, 2008, at 12:36 PM, Douglas Garstang wrote: > >> I'm new to freeswitch. >> >> Is there a vxml parser for freeswitch? >> >> Doug. >> >> >> _______________________________________________ >> 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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