[asterisk-biz] mg3000-r fxo gateway provides more feature to work with asterisk

[GlobalOfficePhone]

Can you provide the link to the website.
I"ll like to see the specs in more details and the whole product range.
Thanks

On 5/10/06, Hao Xu <hao.xu.cn at gmail.com> wrote:
>
>  Hi, every one
>
>
>
> I'd like to introduce some new feature of our products.
>
>
>
> *mg3000-r **fxo gateway **provides more feature to work with asterisk.*
>
>
>
> 1.play asterisk ivr with no interuption.
>
> when the mg3000-r received call from co line, it wouldn't conect
> instantly.instead, it start call to asterisk ivr first,when the ivr ready,
> it connect the co line. this feature make user feel friendly.
>
> 2. pbx voip/pstn inteleged route.
>
> when you make pbx connect to voip/asterisk, how to make voip more stable.
> MG3000-R could detect the voip quanlity, when voip line failed, it change to
> pstn line automaticly.
>
> 3. pstn caller number transfer.
>
> When pstn call in, the mg3000-R start voip call to the asterisk using pstn
> caller number instead of gateway number.
>
> 4. multy region pstn singal support.
>
> By using MindSpeed technology, it integrated many region's pstn singal.
>
>
>
>
>
> *Why we choose a voip fxo gateway while not a asterisk card.*
>
> * *
>
> 1.       voice process is a realtime task. PC operate system is not a
> realtime one. Voip gateway use its own dsp to do voice process while
> asterisk card use PC CPU to do this. Just like the DVD decode, on heavy
> task, voip gateway hardware will do better. So  we sugguest you to use
> voip gateway on more than 4 phone line system.
>
> 2.       Asterisk PC+ voip gateway model, it is easy to expand to over 100
> user. In this scale, you can not plug so many card into one PC.
>
> 3.       There are many analog voip gateway producer, the price is cheap,
> especially for MG3000-R fxs.
>
>
>
> *How MG3000-R work with Asterisk feature.*
>
>
>
> For FXS voip gateway, interoperate with Asterisk is easy. There are two
> requirements: one is sip interoperability. The other is DTMF transfer model.
>
> If  a voip gateway can make call with asterisk, that could to say sip
> interoperability is ok. If the auto attendant service is ok, that is mean
> DTMF transfer is ok.. The other things will be no problem.
>
>
>
> Howerver there are some limits in Asterisk, especially on transcoding. If
> we use g.711, all is ok. but we are normal use g.723 or g.729. when we
> want to use conference service. Asterisk need to change codec to G.711.
>
> The other things you must be careful. When a call setup successfully,
> there need call original voip gateway, and also the call termination voip
> gateway. The interoperation of such two type gateway is also important.
>
>
>
> *How MG3000-R backup voip with the PSTN line *
>
>
>
> MG3000-R 4s4o owned 4 fxo and 4 fxs ports. When you work in backup mode, 4
> fxs ports connect to pbx trunk line, 4 fxo ports connect to CO line. User
> start a voip call from fxs port, When a voip call failed, MG3000-R will
> automatically dial on the pstn line through fxo port. The end user use the
> gateway like it was on voip line. So the voip call quanlity is protected
> with pstn backup line.
>
>
>
> The voip change to pstn condition is: when voip service is unavailable or
> the power is loose.
>
> It is very difference than lifeline. Lifeline is a n to 1 protection. This
> is 1:1 protection. Lifeline work only when the power is loose, it has no use
> when the network is down.
>
>
>
> For more info please contact us hao.xu.cn@@gmail.com
>
>
>
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-- 
Regards,

GlobalOfficePhone
Please note our new email: GlobalOfficePhone.com
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