I have a 3620 with an NM-HDV-2T1 configured with CAS to the DS-1 cards on an Avaya Magix/Legend. The PBX is originating calls to a 5300 connected to the PSTN, and is also terminating (800) calls originated from the 5300.
The issue is that we cannot have inbound and outbound calls configured simultaneously. When we have only inbound or outbound configured on the 3620 it works fine, however when we configure both we get DSP timeout errors.
We have tried all different CODEC's and IOS's to no avail. Does anyone know what the issue might be? Do I have a hardware problem?
Can you paste the error message here? What version of IOS and feature set are you on currently with these errors? DSP timeouts were a bug that was fixed in 12.2(6a). 12.2.13 is latest 12.2 mainline release at moment.
Can you post a sanitized configuration if possible? I've fixed 2-3 problems involving 3600s and NM-HDVs for clients of ours so can help point you in the right direction perhaps. One was code and the other was a couple bad PVDMs (DSP modules).
For me, troubleshooting NM-HDVs I usually use the following commands when placing test calls:
show voice call summary (shows call status). on Inbound calls into NM-HDV the status won't change if the call isn't making it past the DSP
On calls from other sites going out the NM-HDV you'll see a connect message but user on phone probably will get dead silence.
Other useful commands,
test dsp (slot # of controller) then option 1
this will query all the DSPs and they will report ALIVE if they are still up (have term mon if telneted in).
test dsp (slot #) then option 2 (similar to show voice dsp but more detailed) gives you detailed voice-port to dsp id mappings. This way you can pinpoint calls on voice-ports that aren't working to certain DSP id's. Theres a NM-HDV doc on cisco.com with PVDM DSP ID # breakdown per SIMM socket. I have the link at the office.
If you do narrow it down to a flaky DSP you can change the ds0-group # to try to get a new voice-port with a different DSP id. You don't need to change the timeslot #. This will require reconfiguration of dial-peers and voice-ports however. If the ds0-group # isn't a big change you may end up on same set of DSPs. I haven't figured out exactly how voice-ports map to DSP id's yet and have been trying to gather information on this from TAC. It appears to use the first available , but the higher the ds0-group # the higher the dsp id.
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