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

Wed May 10 18:30:41 MST 2006

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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