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Switch to pass traffic.


 Hello i would like to know wich type of device or switch should i use if i have  a person in South Africa and we want to generate calls there  and route those calls through our Equipment to and then we send those calls worldwide.

 It s a specialized switch or its a simple GK , im in the VOIP a few years ago but never try something like this so i would like to know if someone have reference please .

 Thanks

 JC
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juank03
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juank03
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1 Solution
 
erik_nodlandCommented:
Lots of different ways of doing what you want. You haven't been very specific so I can give you a general way of doing this with just a couple of phones.
You need a router both ends connected to the internet. FXS card with an analogue phone in S. Africa. Talks to the router in your main office with an FXO card in the router connected to your PBX which would then send the call out via your PSTN.

Making lots of assumptions here. Give us more detail. Such as number of users in S. Africa. Any existing connections between your sites. Expected number of concurrent calls. etc. etc.
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juank03Author Commented:

 Yeap, i will just let me speak with my people over there i will be back later ok . thanks for your prompt reply..

 Thanks \

 JC
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Joel_SiskoCommented:
JC,

If its for one person, the following gateway supports FXS and FXO.

Sipura SPA-3000 - Single FXS, Single FXO VoIP Gateway. This is designed for what we call "Hop-on Hop-off" applications such as the one you want to do.

This can be found at http://www.voipsupply.com/product_info.php?cPath=3_26&products_id=203&desc=Sipura%20SPA-3000

But sounds like you will have more that that. The Quintum new series of Gateways have many new features and very addfordable. You can implement a fully scalable SIP network with thier built-in dial plans and intergrated Gatekeepers (feature Cisco does not even have).

If you can provide anymore detail of what you would like to do, we can be more specific.

Joel



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juank03Author Commented:

 Ok , it seems that they have customers like call shops and some biz and residentials ok , so they will have devices like ata 186 or any other GW supporting h.323 or Sip and they want to send that to a switch here but i need to know what kind of switch or equipment do u think that i need to route those calls from here to worldwide and send that back


 Thanks.


 JC
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Joel_SiskoCommented:
Client -->Gateway-->PSTN

My question is that will you be terminated the recieveed calls onto the Public Swiched Telephone Network? OR taking that call and terminating into some Internet based VoIP provider?

Will you be able to dictate the equipmnet used at clients site as well as the Gateway?

How many users/traffic do you excpect?
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Joel_SiskoCommented:
Here is one setup:

Tenor ASM200 (client with two FXS and FXO ports) --> Tenor DX8120 (8 port T1/E1/PRI up to 120 VoIP calls in a single box)-->Connect to your PSTN

Also the new Tenor series is all GUI based, esy to manage remotely. They also have Gateways that support a total of 960 calls.
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juank03Author Commented:

 Hey Joel ,

 Where i can learn more about this  Operate my own VoIP network from  PC/Server. Install a GK load it, bring in my own carriers ?

 I will check the specs on this Tenor and see if its enough.

 Thanks.
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Joel_SiskoCommented:
I am not sure how well versed you are in Telecom in general so I will start there. Below is a whole bunch of links, books and info to get things started that are compilation of some of my other posts.

But there are a few good open source systems that can help you out, one being Sip Express Router (SER) and a IP-PBX called Asterisk. I am a firm supporter of SIP, I used to be H.323 but that has changed over the past year.

http://www.iptel.org/ser/     
http://www.asterisk.org/
http://www.voip-info.org/wiki-Asterisk


To help you choose you need to understand where the protocols came from and are going. To start MGCP is around but is the one of the three that is really not deployed much, Cisco is about the only company that does. For various reasons it never took hold and had a chance to compete.

The battle between which VoIP protocols to use has been going on for years. Being that my foundation in Convergence started in the voice industry, I was heavily biased towards the use of H.323, being the standard came from the ITU. In the early days, SIP (from the IETF) was very primitive in its call control and implementation. H.323 is very stuctured and reliable but is a compilation of various protocols from various areas. So its a bit like Frankenstein. SIP is text based and very flexable.

H.323 took a hold a few years ago and seemed to win the VoIP protocol war, the Cisco centric folks think it was mostly part to Cisco, but this is really not the case. The main cause at the time was 911 and CALEA (this enables law enforcement to monitor and record packet based voice networks). CALEA was passed in 1994, but never adheared too for various techinical reasons by the carriers, many extenstions to the time deadline for compliance were passed over the years. But after 911 the goverment could not wait nor extend the deadline for compliance; H.323 was the logical choice at the time for the needs to be able to record and monitor.

Now years have passed and SIP has leapfrogged past H.323 in features and functionality (with the ability to be recorded and monitored). Also, H.323 is so tightly structured that it is very hard  for it to adapt to the fast moving nature of the IP telephony movement.

I believe that SIP is the standard to go with for various technical reasons, but more so that the industry has really geared toward the use of it as the de facto standard. This is evident if you look at the services and products being offered by such companies as Microsoft, Vonnage, Polycom, Cisco and countless others.

Also SIP will allow you to deploy a host of other services for end clients (additional revenues), such as various hard based and soft based phones; deploy additional technologies such as WiFi, Bluetooth and cellular based. By using SIP you can start looking at companies such as Vonnage and other SIP based carriers to terminate or originate calls. Additionally by using SIP you can look at various ASP based applications . SIP is allowing for the VoIP enabling of applications and truly allowing for a Converged Network.

Quick take on VoIP, VoIP is means to transport voice in todays data networks. IP Telephony is the movement to change the traditional telecom market as a whole, which VoIP is a part of, allowing a seemeless and more usefull means of communicating with todays technologies.


http://www.iptelephony.org/

http://www.tmcnet.com/voip/Default.aspx
http://www.dialink.com/

http://www.broadsoft.com/products/applications.html

http://www.ariavoice.com/page/72

http://www.netcentrex.net/

http://www.starvox.com/sp/sp_prd_ov.html

Some areas of study include:

IVR, Voice Mail, TTS, ASR, structured cabling ,BICSI, RCDD, FCC site, CALEA, 1996 Telecommunications ACT

Some books to study:

Newtons Telecom Dictionary, by Harry Newton

Guide to T-1 Networking: How to Buy, Install & Use T-1 From Desktop to Ds-3
by William A. Flangan


Here is a list of links to keep you busy this weekend:

http://www.protocols.com/pbook/VoIP.htm
http://www.protocols.com/pbook/VoIPFamily.htm    basic overview of VoIP

http://www.cisco.com/warp/public/788/signalling/net_signal_control.html      basic telecom link

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080174804.shtml    debug sample

http://www.cisco.com/application/pdf/en/us/guest/products/ps5037/c2001/ccmigration_09186a008011bd39.pdf   all of IOS voice commands

MGCP rfc's below

http://www.faqs.org/rfcs/rfc2705.html
http://www.faqs.org/rfcs/rfc3525.html
http://www.faqs.org/rfcs/rfc3660.html
http://www.faqs.org/rfcs/rfc3661.html 

SIP RFC's

http://www.ietf.org/rfc/rfc3261.txt
http://www.ietf.org/rfc/rfc3311.txt


Just some general phone info:



First the new Toshiba CTX is fully IP enabled, but you get to use your existing phones and do need to change your cables. One of the previous comments talked about Artisoft, they have a decent product, but one issue, few resellers that know how to install a phone system.

The Mitel system is also a very good system, top 3 in the VoIP market today. Only issue with Mitel is the lack of certified dealers this is for various reasons which would take a page to explain.

I have been installing the Cisco product before Cisco purchased it. Works well, but the price point for 150 users is steep and I could not justify the cost.

3COM NBX has a good product, would work on a CAT5 cable plant running 10MB switched. Not keen though on the multi-site features.

Avaya IP office, you think, hey its AT&T, they invented the phone right? Late in the game and have the worst rating for availability out of everyone.

Nortel BCM, great system, great multi-site features, lacks all the other bells and whistles some other systems have, but do you really need a feature called Zoomerang?

Vertical Networks all around great system, but made for the satellite office connecting to a traditional PBX at HQ.

ConvergenceCenter, SIP based system with fully integrated CRM, Business Management software, Knowledge base, IM, and email. Main issue with them is they are a newer company. But so was Ford at one point.


Hope all of this helps!

Kindest regards,

Joel _Sisko






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Joel_SiskoCommented:
JC,

Here are a few links to the Quintum switches, you can review the manuals and see what features they provide:

http://www.quintum.com/support/products/2G/tenor_2G/index.shtml
http://www.quintum.com/support/mgmt/index.shtml


Good Luck!
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