AudioCodes MP-124 Analog to IP

[PhoneGuyy]

I have a question on configuration for this unit...

I got the unit installed and cabled out at the far end, ok no problem. I do get dial tone on the unit, but have questions to what I need to do to get phone numbers from the main site over to this location.

I have 24 SIP port license in my CS1000M rls 5.5 system...

1. What do I have to configure in PBX to make this work
2. What in the Analog Gateway

I have a idea in my head of what I need to do from reading the IP networking NTP, but just trying to clear the fog and make sure..

Thanks for your help...

 

[johnpoole]

you'll need to set up your cdp to send numbers to that route. same as with a tdm route.. just build a rlb pointed to the route the dsc's pointed to the rlb (ld 86 and 87)..

john poole
bellsouth business
columbia,sc

 

[allenmac]

pronei has uploaded a board config file that should get you working with very little adjustments necessary.

Just search for AudioCodes and you should find the thread.

 

[PhoneGuyy]

OK so that is what I was thinking John, I was just trying to confirm my thoughts...

 

[PhoneGuyy]

allenmac...I cant find that ini post from pronei

john - So I dont have to program numbers in the NRS

 

[allenmac]

The thread with the board config file is: thread798-1465329

You will still need to program the DNs in NRS so that the NRS can provide IP address resolution to the AudioCodes endpoint.

 

[PhoneGuyy]

ahh hahh.....

So I lets just say I have a DN of 5570 so I have to program that in LD 87 as a DSC etc pointing to my virtual route and also put that number in the NRS(signaliing Server)

 

[allenmac]

Yes, you would add it as a routing entry with cost 1 under the AudioCodes endpoint.

 

[PhoneGuyy]

Ok the foggy in my head is starting to clear up a bit now...

 

[allenmac]

In a nutshell

Inbound call:

1. a call comes into the system destined for one of the ports on your AudioCodes device.
2. Build that dn in CDP as a DSC using an RLI that points to your SIP Virtual Route.
3. PBX sends the call to the NRS.
4. The DN is programmed as a routing entry against the AudioCodes endpoint.
5. The NRS sends the call to the AudioCodes device.
6. The AudioCodes device accepts the call and delivers it to the proper port.

Outbound call:
1. An analog device on one of the AudioCodes ports goes off-hook, and dials a number.
2. The AudioCodes checks it's dialing table to determine how to handle the call. (Send it to the Proxy/Redirect Server)
3. The AudioCodes passes the digits to the NRS for validation.
4. The NRS checks it's routing tables to see if it can find an endpoint for the call.
5. The NRS either rejects the call, or sends back a list of endpoints that can accept the call.
6. The AudioCodes routes the call to the 1st endpoint on the list.
7. The Call is routed to a PBX over it's SIP virtual route, and is routed to either an internal DN, or is routed based on call type via CDP/UDP/NARS/BARS, etc.

 

[PhoneGuyy]

OK so I understand the process that needs to happen for the calls to work...

The part that I am lost on because I have not done before is programming the NRS....

Does anybody have any screen shot or something they could send me to look at to see what kind of info goes in to configuration?