PSTN vs ISP Redundancy

[asingh005]

HI,

I want to know that in VOIP we buy a pstn connection for redundancy in case if our wan connectivity goes down, call flow can go from the pstn connection .But why do we need do we need a pstn connection ? why don't we go for another ISP connection for redundancy ?
I'm new to VOIP . Please help as i'm keen to learn

Regards 
Aateek singh

[Terry Cheema]

If you have Parent Nexus5k/7k then you can look at Nexus 2K switches,

http://www.cisco.com/c/en/us/products/collateral/switches/nexus-2000-series-fabric-extenders/data_sheet_c78-507093.html

-Terry

[asingh005]

Hi Terry 
I failed to understand the document for Nexus 5k/7k .Can you help me to understand ? as i'm new in networking .
I only got to know that we can use nexus2k as a line card to nexus parent 5k .
i just want to know that if i want 96 GigE ports in cisco switch then i have to stack a switch with other so how to go from here now ?Means which switch i have to select 
Please help .  
http://www.cisco.com/c/en/us/products/collateral/switches/nexus-2000-series-fabric-extenders/data_sheet_c78-507093.html

[Terry Cheema]

Yes Nexus 2k is option only if you have existing 5k/7k in your environment. Regards to stack as per my previous post if you get two switches, lets say two 3850 48P switches, you can connect them via stack cable and its converted into a one logical switch with 96 ports.

Hope its clear.

-Terry

[asingh005]

Yes Terry its clear :)
Yesterday i first understand what is stacking and then read your post.
Thank you.

Regards
Aateek singh 

[Terry Cheema]

No worries Glad that helped.

-Terry
 

[asingh005]

Hi terry could you please tell me 
what is sip,h232,voice gateways,Pri lines and DID. 
I have read documents but sooner or later i always forget about these may be because i fail to understand.
Could you explain me the basics by sharing your experience ?So that i'll understand 


Regards
Aateek Singh

[Terry Cheema]

Voice Gateway is usually a router where your PSTN lines terminate (FXO/BRI/PRI). As the name implies its the gateway for the voice Calls to/from outside your organisation.

PRI lines is an E1/T1 ISDN line. As said it terminates on your voice gateway and connects you to Telco. Over this line external calls are received/sent.

DID or Direct Inward Dial is the block of numbers you would buy from your Telco. The DID are ties to your ISDN lines by Telco programming. Any time someone calls those numbers, the Telco would receive the call and forward it to your voice gateway/ISDN line. Which is then normally translated to a 4 or 5 digit extension by the voice gateway or the Call Manager.

SIP, H323, MGCP are the protocols used for communication between the Voice Gateway and the Call Manager. You can select any of them in your deployment according to the needs.

Hope its clear.

-Terry

[asingh005]

Thank you terry :)


Regards 
Aateek Singh

[asingh005]

Hi Terry,
I was reading about Cube and sip trunking .And came to know that CUBE is a feature in a router with supported ios .So if i get a SIP trunk from my service provider pointing toward my CUBE and from CUBE i distribute these to other customers .

I got stuck here first of all if my customers are having IP phones connecting to IP pbx in there side only so will the communication will happen ?

if i point my Cube toward there side with outgoing and incoming dial-peer ?
in this scenario does i require a cucm or cme or not ?

please help
 

Regards 
Aateek

[Terry Cheema]

Hi Mate,

Yes first of all you are correct CUBE is a functionality that can enabled/disabled on the routers (ISR/ASR).

I am not sure if I correctly understand your question but I will try to answer it based on the information above:

CUBE is just a gateway, you can have a SIP trunk from your provider with a block of  DID numbers. So any incoming calls to that block of DID numbers will be routed to your CUBE. Now it's up to you where you want to send the outgoing calls to - any destination IP PBX calls as long as you have IP connectivity to that IP PBX and the far end is configure to accept the incoming calls.

But having said that I am not exactly clear on what you are trying to achieve. If you can explain it a little more on your call flow would be more beneficial. What is the current call flow and what are you trying to achieve and what are your expected calls flows.

-Terry

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