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Most of that document has changes made in the webpage. The webpage isn't working because the certificates are expired. At the end of the document it talks about using SMnetSetup to change the date and time in order to renew a certificate. This is what I was looking for. However the...

I have a customer with an old (6.2) System Manager. Its certificates have expired, and the normal procedure to renew them no longer works. I was wondering if there is any way to set the date back to before the certs expired so that they can be then renewed. I can find no information on how to...

The 1049 has to do with the fact that these are tie lines, and is on all calls that come across the PRI. I tried routing the 988 calls to the 800 number, and that worked.

I am having a similar issue, but with calls tandeming off of of a CS1000M I have set things up identical to how they are routing 911 and 9+911, and those calls route fine. The d-channel message is the same, except the 988 calls fail with the following message: DCH 21 IMSG SETUP REF...

I have a customer that uses analog stations on a 911 console for non-emergency calls. These stations are members of a DDC hunt group. The customer was recently upgraded from 7.1 to 8.1. After the upgrade, calls that are transferred to the hunt group lead number do not show the correct...

The only thing you lose is that after 30 days you cannot access the system via SAT. If you reboot the system your 30 days starts over again. I know people who do this with lab systems.

I have a customer which is a school district with a virtualized CM and G450 and G430 gateways in the individual locations. They are on 8.1. They have an issue where pages are disconnecting after exactly 2 minutes. The customer has a Bogen P135 Intercom system, integrated with the CM using a...

I have a customer with a CS1000M Single Group (Option 61C) and a CS1000E chassis, both on 7.6. There is a T-1 between the two locations that is being used as a data pipe. On each end is a SR10004. This has been in place and working for years. They are using Session Manager for network...

Turned out that the new gateway had a 255.255.255.0 subnet, and the existing gateways were 255.255.254.0. When I fixed that the problem went away.

The reason we put in the new gateway is that the customer is going with SIP, and the SIP carrier told them not to send fax traffice over the SIP lines. I set up the new IPNR in order to do location based ARS routing, which works out of their RightFax server, but not the analog fax lines. This...

I have a customer who is a hospital. At the main location they have two G450's with a mix of analog, digital and PRI cards. Most of the phones are IP. They wanted to add a G430 with two MM711 cards for fax machines. Gateway is up and cards have dial tone, but when they call outside numbers...

A customer of ours reported all stations on a MM716 dead. When I pulled the board, a single port on the circuit board was burned. There was another borad onsite with the same port burned. I am guessing that we replaced this board once already. There is obviously an issue with the station...

Thanks. That pointed me in the right direction. It is weird, though. On the 8.1 system I was working on I had to reverse the last two parts of the command. It would up being: add abbreviated-dialing enhanced 0 list 1 and add abbreviated-dialing enhanced 1 list 1. (I am using 3 digit...

I have a customer who wants a "System" list of 150 entries. I know that Enhanced Abbreviated Dialing should allow this, but when I create the list it only lets me set the size up to 100. I have verified that Enhanced Abbreviated Dialing is enabled, and that my code size is set to 3. Any ideas?

The incoming format had to be blank for it to work. That was the last change I made, per a document from Avaya I found. I don't work directly with the AVST, so I don't have that information.

I figured it out. Had to fiddle with the fields in the tandem calling table, but it now seems to work. Here is my table, in case anyone wants to know what I did: CALLING PARTY NUMBER CONVERSION FOR TANDEM CALLS...

That makes sense. However, when I tried it, I am still getting +1<voicemail ext> on my caller id. I have never used this table, and am not exactly sure how to program each field.

AVST has looked at it. they are sending <voice mail extension>@sip domain, which is what they say they are supposed to send. When I do a list trace I see the originating extension as <voice mail extension>, which is also in the Public Unknown Numbering table with the 10 digit number associated...

AVST says the issue is with the CM. Avaya Session Manager group says the issue is with the CM. Avaya CM group is not being responsive, and don't seem to have a clue.

This was first noticed when the customer wanted to use the AVST mobile app, but it can be reproduced using the AVST config menu to generate a test call.