June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Is freeswitch can call mobile phones?
by Edmar Cruz 19 Jun '09

19 Jun '09

Hi is there any possible free sites ip that i can connect so I can could to any mobiles phones? I know some several ip sites has the capability to call for free Ip to Voip... I know freeswitch can do this Can you give me an example site? -- View this message in context: http://www.nabble.com/Is-freeswitch-can-call-mobile-phones--tp24088222p2408… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Bootable ISO Daily FreeSWITCH build
by Kristian Kielhofner 18 Jun '09

18 Jun '09

Hello everyone, I've setup one of my build servers to do a fresh check out of SVN trunk and build AstLinux with it every day at 2AM EST. The ISO and build log (for the curious) are available here: http://mirror.astlinux.org/freeswitch/daily/ I just ran a test build but daily builds will begin showing up this evening/morning at 2AM. I plan on keeping about 30 days worth of ISO images. They should be bootable on just about anything including VMware and various other virtualization platforms. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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Voice lag in conference
by Bradley Brashier 18 Jun '09

18 Jun '09

I'm creating a conferencing product for use in a system with theoretically several hundred concurrent calls. I'm using FreeSwitch to create this product, but am not only new to FreeSwitch, but also the entire telecom industry as well as Open Source projects in general (I'm a recovering BIOS guy). I've got a bare-bones conference up and running on the server, including a handshake and a couple of features, and am using the default packages from the current trunk, but I've noticed that voice lag is a pretty big issue. Common lag times are several hundred milliseconds, and I've heard as long as a second. It seems to be at least marginally specific to individual phones -- certain phones have longer lag than others even on the same call. My question is really about what my options are. Is this just a part of SIP? Of conferencing? Of FreeSwitch? Are there things I can prune or slim down that will help? Is this a common issue? If it's common, is it expected by the marketplace?

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Buy The FreeSWITCH Developers Dinner!
by Michael Collins 18 Jun '09

18 Jun '09

Hello FreeSWITCHers out there! I have it on good authority that the FreeSWITCH developers have all convened in an undisclosed location. Rumors that they are plotting to take over the world are not yet confirmed but I will keep you updated as information becomes available. :) It would be great for all of us to show our support and appreciation to the guys for all the hard work they've done. How many of us have had a question answered on the IRC channel or here on the list by one of the guys? How many of us use FreeSWITCH every day for work? If you've benefited from their hard work then please give a little. If we can get everyone to hop on the paypal link (on http://www.freeswitch.org) and donate $5 or $10 then we can easily pay for a nice dinner for the guys. Please hit the link and let me know (off list) when you've donated. Let's do this, people! -Michael

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Sample rate and recordFile
by Andy 18 Jun '09

18 Jun '09

Hi, Sorry but I just can't find this in the documentation. I'm using recordFile to record incoming messages. I'd like the audio files produced to be 11025Hz rather than 8kHz is this possible? What setting do I need to change? Many thanks Andy

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call quality problems in conference
by Victor Toofic 18 Jun '09

18 Jun '09

Hi all! I'm having some troubles with call quality using conferences. The scenario is like this: An agent makes a call to freeswitch and enters in a conference room waiting for outbound calls; on the other side there is an application generating outbound calls and when one is answered it is assigned to the first agent available, so the outbound call enters in some of the agents's conference room (it is some kind of semi-predictive dialer). I'm using conferences because we need special features like monitoring or whispering to the agents. There are times when some of the outbound calls that enter in a conference room have really bad quality: broken/choppy voice, echo, etc, Something like this: http://www.voiptroubleshooter.com/sound_files/40pct_rand_plc.wav This occurs in 20%-40% of the outbound calls. I know it might be because of the jitter or packet loss with our voip provider. But.. this hardly occurs when the agents dial manually (using the bridge app); when dialing manually the problem (when it ocurrs) is always unperceptible. Thats why I think the conference room is aggravating the problem. Im using the 'jitterbuffer_msec=180' in the originate command and the same in the dialplan (when the agents log-in). What do you think is happening here? Am I missing something? Any guidance will be really appreciated! Thnks!! -- Regards.. Victor Toofic

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Re: [Freeswitch-users] VAD, TALK and NOTALK events (Brian West)
by Steven Brown 18 Jun '09

18 Jun '09

Thanks Brian, Yes I had been looking for TALK and NOTALK, CUSTOM conference::maintenance works great. Steve > > > Message: 4 > Date: Thu, 18 Jun 2009 08:16:58 -0500 > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] VAD, TALK and NOTALK events > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <866F2582-2968-4A33-9B1E-CCBBE6294FBA(a)freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > I suspect you're going for TALK and NOTALK as the event names? > > its CUSTOM conference:: maintenance > > /b > > > > On Jun 18, 2009, at 8:00 AM, Steven Brown wrote: > > > Hi, > > > > I have been trying to pick up TALK and NOTALK events but with no > > success, I have enabled VAD for "both" in my config and the rtp is > > stopping and starting as expected however when I hook up to the event > > socket and request "event talk notalk" nothing is ever fired, any > > thoughts on where I am going wrong appreciated. > > > > Thanks > > > > Steve > > > > -

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Which GSM gateway to buy?
by Diego Viola 18 Jun '09

18 Jun '09

Hi everyone, Can you please recommend me some GSM gateway? I'm currently looking for a good one to buy... anyone have experience PORTech GSM gateways? Are they good? I also need it to work with FS, I'm also kinda new with VoIP hardware. Thanks, Diego

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Automatic call distribution
by selva kumar 18 Jun '09

18 Jun '09

Hi, I have setup FS for both inbound and outbound.It is working fine. Now I would like to configure Automatic Call Distribution(ACD).How to configure it in Freeswitch? Sam

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Use Freeswitch to provide a SIP Video/Voice email service
by Bilbo 18 Jun '09

18 Jun '09

Hi, I would like to use Freeswitch to provide a Video/Voice mail service that is integrated with an email service. I would like to have the ability to email the Video/Voice messages as well as the SIP users being able to collect their Video messages using their video soft-phones. Has anyone done this before or know if Freeswitch is capable? Thanks -- View this message in context: http://www.nabble.com/Use-Freeswitch-to-provide-a-SIP-Video-Voice-email-ser… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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