Re: [asterisk-users] DS3 Interface

[zoachien]

I don't see how this is relevant to the discussion.

Zoa

Matt wrote:
> http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
>
> On 10/9/07, *Brian West* < [EMAIL PROTECTED] 
> > wrote:
>
> You apparently don't realize you're talking to.  Thats ok,  You
> keep working on it from your angle.  We are evaluating when the
> time is right to implement this.  We aren't doing this for
> Asterisk we are doing it for FreeSWITCH.
>
> /b
>
> On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote:
>
>> Competition is a good thing.  Let's say you fail or your
>> implementation 
>>
>> is not as robust as the other project or visa versa.  Just as
>> long as 
>>
>> the hardware vendor is different, it should be a good thing.  If
>> it the 
>>
>> same hardware vendor, then maybe you two should work together.
>>
>>
>> Thanks,
>>
>> Steve
>>
>
>
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Re: [asterisk-users] "Click to Talk" Web Applications with Asterisk

[zoachien]

Google for mexuar.

Zoa

Anselm Martin Hoffmeister wrote:
> Am Dienstag, den 09.10.2007, 14:23 -0500 schrieb Ricardo Melendez:
>   
>> Hi, I would like to develop a “click to talk” app to interface with
>> asterisk, anyone know about some SDK/frameworks to implement this.
>> 
>
> I have not ever used such an application, but there are several
> solutions commercially available. If your intention is getting a
> solution, you might consider spending money. If your intention is
> learning, the better - but sorry, I cannot give adequate pointers there.
> I remember there were open source puzzles parts that could be mended to
> something like a web click-to-call app, might be the term "jiaxclient"
> relates to that. Do not count to much of that, my brain is getting old.
>
> I do not want to advertise a specific solution, but you could search the
> mailing list archives - "click to call" might be a subject worth
> reading. You could also look for something like "IAX Client JAVA". I bet
> there is also some information to be found on voip-info.org. I think at
> least one vendor offers free trial versions so you could at least test
> wether the concept is viable, and then decide to either spend money or
> time on the project.
>
> I hope you did not trigger one of those "Hey, I have a solution for
> you", "hey, this is a non-commercial-list", "go die" flamewar - we had
> enough of those ;-)
>
> Best regards,
>
> Anselm
>
>
>
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Re: [asterisk-users] HOWTO/FAQ question (Location: Sweden)

[zoachien]

Turbo Fredriksson wrote:
> Sorry for this. This is most likely a HOWTO or FAQ question, but
> it's so much information and documentation to wade through so
> I hope someone could take a minute to answer anyway.
>
> If not, no worries. I'll get to it sooner or later :)
>
>
> I'm trying to understand what Asterisk actually is and the basic
> workings... I think I've understand what I need to get going,
> except one thing.
>
> How do I connect to a 'normal' (i.e. analog) telephone? That is,
> if my company/project have 100% IP telephony, but one of these
> phones need to call a analog telephone in another company (or
> if I need to call home for any reason :). What do I need from
> the 'phone company'? And what hardware?
>
>   
You need something to interconnect to your telco, this can be done on 
different ways:

- you can take a voip provider and not buy any hardware.
- you can use a TDM card and connect to a classic analog telephone line. 
(1 simultaneous call per port)
- you can use a BRI card and connect to a classic isdn line (You get 2 
lines per port this case)

If you need more capacity, (more than 2 simultaneous channels), just 
think PRI or PRA or E1 as one and the same thing. (30 simultaneous calls 
per port).

Zoa.

> This is Sweden with Telia as provider if that matters.
>
>
> DISCLAIMER1: I've read about BRI, PRI, E1/T1 etc, but the reason
>  I'm all confused is that they (Telia) don't seem to
>  understand the question, and also claims that a
>  PRI == E1.
> DISCLAIMER2: I've seen the Digium cards, but due to confusion with
>  Telia, I'm not sure if I want/need a Digital or an
>  Analog card... And 'how big'...
>
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Re: [asterisk-users] Queue stats

[zoachien]

Google for OrderlyQ or Queuemetrics. (in random order)

Zoa

Scott Wolfe wrote:
> What do you want? Maybe I can write it into ASTassistant.
>
> Scott
> http://www.astassistant.com
>
>
> - Original Message - 
> From: "Matt Riddell" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Wednesday, August 29, 2007 2:35 PM
> Subject: Re: [asterisk-users] Queue stats
>
>
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Ed Nuñez wrote:
>   
>> Can anyone recommend a good commercial solution for queue statistics?
>> 
>
> http://queuemetrics.loway.it/
>
> - --
> Kind Regards,
>
> Matt Riddell
> Director
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Re: [asterisk-users] Gizmo revisited

[zoachien]

randulo wrote:
> On 8/24/07, Carlos Leal <[EMAIL PROTECTED]> wrote:
>   
>> Launched the OS X version of Gizmo after about a year of inactivity,
>> downloaded the update and discovered the new improved Giszmo features
>> Asterisk interoperability by allowing a secondary SIP account to be
>> registered simultaneously.
>>
>> It also allows you to make the routing choice for outgoing calls;
>> your own server or via Gizmo. So far, this is the best SIP softphone
>> I've come across for OS X.
>>
>> It comes in other flavors and I thought I'd mention it as it can be
>> free and I haven't seen it mentioned recently.
>> 
>
> Yes, I agree, this is a darn good free SIP client with a few features
> such as recording, playing WAV files (good for podcasting, or if you
> want to irritate people you call with laugh tracks or something).
> I don't find it better than X-Lite, but about the same. If I had to
> choose one client, it would probably be Zoiper, because it does SIP
> and IAX well (or at least I've had good results with it). Zoiper also
> has a very small footprint on the GUI.
>
>   
Cool, thanks :) (we are making that zoiper phone).
Carlos, i' already working on the gui design for version 3.0, if you 
send me your comments ([EMAIL PROTECTED]) on what you dislike in 
ours or like in other phones, i will certainly have a look at your 
comments to make it better.

Zoa
> /r
>
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Re: [asterisk-users] IAX2 trunking scalability

[zoachien]

I would suggest to use the branch with the iax bug fixes, i trust it 
more than the stable version.

Zoa

Jean-Michel Hiver wrote:
>> I used to do it, but its a while ago. (Before iax2 got some more fixes)
>> The trick was to keep the trunks small (like 40 per trunk, just make
>> multiple), this should no longer be needed.
>> Cpu utilisation with trunking should be lower than without trunking.
>> 
>
> Hi Zoa,
>
> Thanks for your input. I think I'll set up two boxes and do failover  
> + loadbalancing, just in case one box decides to crash =)
>
> Cheers,
> Jean-Michel.
>
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Re: [asterisk-users] IAX2 trunking scalability

[zoachien]

Jean-Michel Hiver wrote:
>> So you are using an asterisk box as an E1 gateway. You want to know if
>> switching from not using IAX trunking to using IAX trunking will have
>> any effect? Yes it will lower your bandwidth usage a little. It
>> will not increase the CPU load. If your system can support x calls it
>> will be able to support the same amount of calls.
>> 
>
> On about 1/2 E1, it shows that bandwith usage has been about halved - i.e.  
> without trunking each G.729 call takes 50 kbps (inbound + oubound) and  
> with IAX2 trunking it takes about half of that (using trunkfreq=40). Which  
> is good!
>
> I'm wondering wether anybody already had a IAX2 trunking ON and managed to  
> push 3 E1s worth of traffic without issues.
>
>
>   
I used to do it, but its a while ago. (Before iax2 got some more fixes)
The trick was to keep the trunks small (like 40 per trunk, just make 
multiple), this should no longer be needed.
Cpu utilisation with trunking should be lower than without trunking.

zoa
>> The best thing you can do for your system is add a TC400B card. It
>> will also legally support G723 codec which I think sounds just fine,
>> but will save you a bit more bandwidth. Using the hardware transcoder
>> will greatly increase the number of calls your system would be able to
>> handle.
>> 
>
> I'm already receiving the calls as g.729, so there is little gain  
> (slightly less bandwith usage, slighly worse sound) in doing g.729 ->  
> g.723 transcoding - while doing IAX2 trunking vs NOT doing it seems to  
> half the bandwith requirement.
>
> Cheers,
> Jean-Michel.
>
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