06-13-2010 09:51 AM - edited 03-13-2019 07:00 PM
I currently have one incoming T1 based NI2 PRI which terminates into an Adtran Atlas which then generates two network-side PRIs. The Atlas routes incoming calls between the two network-side PRIs based on DID numbers. I want to remove the Atlas from service and move this functionality to my 2821 router. My 2821 has the VWICs to do this but no DSP modules and I'm hung up on what I really need re DSPs. Since I currently don't want any call to end terminate on the 2821, do I need DSPs modules? All PRI switching configuration examples I've read involve some sort of VoIP operation which I don't need to deal with right now.
-mick
Solved! Go to Solution.
06-13-2010 06:03 PM
It doe not matter which type of call it is, you still need DSP for the simple fact you are configuring a voice interface.
check:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00804794c6.shtml
06-13-2010 11:51 AM
What do you want the ISR to do with the calls then?
06-13-2010 02:55 PM
SImply route them from the incoming PRI to either of the two network-side PRIs. This is an operation the Atlas refers to as digital cross connect.
-mick
06-13-2010 03:13 PM
The closest that Cisco has without involving a call agent and DSPs is called drop-and-insert:
Cisco IOS Voice Port Configuration Guide 12.4T
...Some two-port T1 or E1 VWICs can provide drop-and-insert multiplexing services with integrated DSU/CSUs. For example, when used with a digital T1 packet voice trunk network module, drop-and-insert allows 64-kbps DS0 channels to be taken from one T1 and digitally cross-connected to 64-kbps DS0 channels on another T1. Drop and insert, sometimes called time-division multiplex (TDM) cross-connect, uses circuit switching rather than the digital signal processors (DSPs) that VoIP technology employs.
What you appear to be asking is for the router to make the route decision after evaluating the Q.931 ISDN SETUP PDU which specifies the DNIS. As far as I know, this will require PDVMs and a call agent (i.e. dial peers).
FYI here is the new DSP Calculator: http://www.cisco.com/web/applicat/dsprecal/tdm_services.html
06-13-2010 03:38 PM
I had done some reading in the areas you posted.
a) The drop and insert feature (as far as I can tell) is only for hard wiring fixed T1 DS0 channels and not dynamic PRI B channels.
b) Your description of evaluating the PRI D channel Q.931 info (for "Called Party number") is accurate.
c) When I had used the DSP calcluator, it seemed to focus on codec selections assuming VoIP applications not pure "passthrough" of whatever is coming in on the B channel - which in my instance could be a voice call, modem or fax call or an end-to-end 64k ISDN call.
-mick
06-13-2010 06:03 PM
It doe not matter which type of call it is, you still need DSP for the simple fact you are configuring a voice interface.
check:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00804794c6.shtml
06-13-2010 06:20 PM
Perfect. Thats the doc I needed to find - thanks very much!
So my gut feel that I didn't need the DSP for the duration of the call is right but then again its moot if I need one for call setup.
Guess I just got in the market for a pvdm2 module ;-)
Thanks again.
-mick
06-13-2010 07:00 PM
Thank you for the nice rating and good luck!
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