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DSP req help needed

hi all

i have the following scenario:

single site ip telephony with 2 7825 ibm MCS's Pub & Sub

users are 160 users

there r 2 redundant 2821 gateways each one has the following:

2 ports NM-HDV2-2E1


PVDM2-32 included

C2800NM-SPSERVICESK9-M Version 12.4(3g), RELEASE SOFTWARE (fc2)

ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1)

System image file is "flash:c2800nm-spservicesk9-mz.124-3g.bin"

there are 2 E1's PRI, one E1 on each gateway

my question is on DSPs required for voice termination and conference,

is PVDM2-32 on each gateway enough??

i ask this because when i added the command:

Pri-group timeslots 1-31 service mgcp

it gives me an error like that no dsp enough for 30 channels, it is only enough for 28 channels?!

now, when i added a phone and configured the gateway,and made a call from pstn to this ip phone it rings but i did not hear ringback tone, and when ip phone went offhook there is no voice heared at all!!

does any one have an idea how to solve this problem, i think the all story is about DSP's?!!

attached is the output of sh dspfarm all

Hall of Fame Super Silver

Re: DSP req help needed

PVDM-32 is not sufficient for 2 E1s and 4 FXO, using G.711 you will require 30*2+4=66 channels for voice termination only. If you use G.729 you will need to add 50% on top of it. I suggest you run the calcuations through DSP calcuator:



New Member

Re: DSP req help needed

thanks Chris

if so, then i have 32 channels to terminate 32 g711 calls, but why is the output for the sh dspfarm all


Total number of DSPFARM DSP channel(s) 0

and when calling from pstn there is no ring back tone and there is no voice heared at all neither from called ip phone side nor from caller side??

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