login | register
Wed 01 of Aug, 2007 [09:27 UTC]

voip-info.org

Search with Google
Search this site with Google. Results may not include recent changes.

Web www.voip-info.org
Shoutbox
  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.34s
  • Memory usage: 2.18MB
  • Database queries: 31
  • GZIP: Disabled
  • Server load: 2.04

MG3000-R

mg3000-r fxo provides more feature to work with asterisk.

Feature
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 hao.xu.cn@@gmail.com

Created by hawkxu, Last modification by hawkxu on Wed 10 of May, 2006 [08:43 UTC]

Comments Filter

Who make it ?

by Li Zheng on Wednesday 10 of May, 2006 [08:08:59 UTC]
Is MG3000-R a Gohigh product?

Thanks.

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:

Terms of Service Privacy Policy
© 2003-2007 Arte Marketing, Inc.

Powered by bitweaver