The problem is that the Asterisk users ARE NOT IP Office users, so the IP Office does not recognize them all, I would try redirecting Asterisk calls to MADN/phantom users which are forwared to the PRI (never tried TBH).
Anyway if the TouchToneTommy suggestion works I think 50 users is not such a...
If it's a new card in a empty slot you just have to insert it an power up the system; the PRI card will be automatically recognized and enabled.
Viceversa if the slot is actually occupied by another daughter card I would recommend the following steps:
Power off the system then remove phisically...
H.323 firmware is allowed on specific models only but, anyway, the problem remains 'cause even if in H.323 they won't be recognized as Avaya devices, you'll need a 3rd pty ip endpoint license.
Nope Rhino, the system does not recognize at all the whole series, it means you can't use the Avaya IP Endpoint licenses, even if available in the system
I suppose your have a R11.0 release on your Server Edition. Basic Users do not need licenses at all, this is SURE, if my memory is not collapsing that was a well known bug on the early R11.0 System Status application, you can check it all easily by installing the latest Manager Suite, anyway I...
I suppose you mean for group and individual mailboxes on incoming calls...
First of all you should start from a wav file (no MP3) edited as 8KHz, 16 Bit, Mono. MP3 is not allowed and ripping a wav from a MP3 neither because MP3 is a lossy format.
Then you can select the LVM Greetings utility...
Assuming you are using XIMA or Avaya ACR, if the agent is already on a call the call center should not route the call on it, it's a non-sense in terms of service, the Agent is formerly busy on statistics.
You should edit the 46xxsettings.txt file, have a look at the standard one and you'll find out how to, then add those strings to the autogenerated one (you must replace it)
The IPO behaviour is a typical antifraud mechanism, the IPO doesn't allow outgoing traffic on SIP trunks if generated by "non-ipo users". Anyway i remember someone else here in the past faced this issue and some one gave a trick, may be the "PAI" setting in the SIP credential page but I am not...
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