The gateway configuration and CUE configuration isn't very helpful because this is all MGCP controlled.
I would suggest opening a TAC case so you can pull traces and find out what's going on.
Dear Mr. Nick,
Thanks for your reply. Is there any other troubleshooting tips before i open a TAC case. I will be very thankful to you.
Awaiting for your reply.
I would do some test calls and track them with 'show voice call summary' and look at exactly which port you're coming in. If you can figure out how to recreate the problem, it can be solved many times faster. Right now, the problem description isn't very clear and it's very muddy to troubleshoot.
Unfortunately the outputs are too large for something to look at on a web forum.
I would work on figuring out exactly what the problem is, and if you can create a more clear problem statement, we will be able to better help you.
A clear problem statement would be like this:
CUE - cue auto attendant script intermittently working.
Like for example when somebody dials the PSTN number the AA sometimes picks the call and sometimes it does not picks the call (its keeps ringing)
Second thing is that sometimes the if you call the PSTN number, the called is picked up but there is no response from the AA.
Unfortunately even for a problem such as that, you will need to run a lot of debugs for a large amount of time (since you can't easily reproduce the problem)
That would be best handled by TAC.
It should be noted that if you have a custom script on the AA, and it's CUE, then TAC will not help you support the script. They will make sure that the call is getting handed off correctly, and after that the burden is on the writer of the script.
On the 3845, try entering the following
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
Hope that helped, if so please rate.
Dear Mr Kenneth,
thanks for the reply...i will add these commands and will let you know about the result..thanks for your suggestion.
Dear Mr Kenneth,
I added the above mentioned commands. But still no luck. Any other suggestion will highly be commendable.
it sounds from ur AA secripts if you have aCRS try to debug it and see if it has problem with the answer step or what ever after it
Despite the fact that this post is very old, but I thought the creator didn't achieve what he wanted using the translation and ended up using corlists. Anyway here is how you can achieve it:
Assuming you are using 9 for outside dialing:
Voice translation-rule 1
Rule 1 /^9\(. *\)/ /19\1/
Voice translation-rule 2
Rule 1 /^9\(. *\)/ /29\1/
Voice translation-profile phone1
Translate called 1
Voice translation-profile phone2
Translate called 2
Translation-profile incoming phone1
Translation-profile incoming phone2
Dial-peer voice 1 pots
Dial-peer voice 2 pots