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
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?!!
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:
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