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So I have a Centralized 6.3 CM with several sites running off of it with just SIP phones. Gateways are only located in the data center not at the sites. The sites span 3 states and the desire is to play a separate music on hold file by state. Where would be the best place to do this? Separate...

So based on the PSN the only thing affected would be Signaling Servers or Co-Res SS/CS that are 7.6 pre service pack 6 or 7.5, 7.0 or 6.0 correct. The vxworks call servers wouldn't be affected. Are you disabling ntp on the signaling servers then? If so how?

I figured it out. Issue was due to my test extension range not being in the Dial Plan Analysis. (Sorry old Nortel guy) So what I am going to do is add 1 through 9 to the Dial Plan analysis as a 4 digit length and making it call type UDP. Then in UDP 1-9 will be an entry with L3 to apply the...

For testing purposes I put a location prefix in my test location. I used 999. So dialing a 4 digit number that should convert to 999-1234 correct? We do not use an access code for ARS 1 is defined in uniform dialing plan to send to ARS so no match to an extension sends the call to Session...

Tried the change calltype analysis on my test lab CM and it doesn't seem to make the change. The list trace of the station shows I'm in the right location it just doesn't seem to recognize the 4 digts. I'll keep trying. Thanks gwebster for the recommendation I had read a bit of that document...

Looking for some assistance with setting up short dialing for location based routing. CM 6.3 96x1 phones with SIP firmware SM 6.3 We are implementing a single CM with SIP phones only to replace our CS1000 and key systems at 100+ sites. Our dialplan is 11 digit extensions so every phone...

I have been investigating how to do this myself. Using System Manager it works but your credentials don't follow to the CM if you want to use ASA or Putty. I found this document a while back but haven't tried it yet. It's on my to do list. https://downloads.avaya.com/css/P8/documents/100060122

Thanks for the input. Any idea how many entries are allowed in the public unknown numbering table?

My system CM6.3 with SIP 96xx phones SIP trunks from CLINK no ISDN or POTS lines connected. CM serves approximatley 280 different sites I'm trying to figure out a good way to change the outbound caller ID on a few sets on a per site basis. We have a small group of people that make outbound...

So I took the second Coverage Path off the phone and set the 1st coverage path to DNC/SAC/Goto Cover is y for both inside and outside and the feature works. So for some reason it's not recognizing that there is a second coverage path on the phone. I opened an Avaya case too.

Failed to my cell phone as well. Here is the trace. I changed the phone numbers to bogus ones. 09:08:05 TRACE STARTED 10/16/2014 CM Release String cold-03.0.124.0-21291 09:08:11 SIP<INVITE sip:17003991234@sip.abc.com;avaya-cm-fnu=of 09:08:11 SIP<f-hook SIP/2.0 09:08:11 Call-ID...

Yes the CP still exists. I have gone back through the history and have not found any changes yet but we are building a log of phones so it could be lost in that list somewhere. I don't know of any system parameters that have been changed though.

Strange issue here. This worked for a while and then stopped. Setup CM 6.3 AAM 6.2 SM 6.3 SIP 9611 phones We have several sites connected to this CM. The sites have main numbers that ring to a coverage path that contains a cover answer group for their site during the day and AAM at night. When...

Sorry for the lack of details no offense taken. I posted this like an idiot so lets start over. We are CM and SM Release 6.3 using SIP 9611 and 9601 IP phones. 9611 Phones are upgrading from 6.2.1 to 6.4.1 firmware. 9601 phones are Release 1? trying to go to 6.2.1 then to 6.2.4. The phones...

Has anyone come across a way to get the phone out of this loop? On some of the phones that are doing this we can press * to get into the basic config menu on the others we can't.

Thanks for the information. I'll open a GRIP(E) through our account team as well.

We created a way for our users that answer a lot of calls to be able to transfer someone directly to another users voicemail box on our AAM. Our users are confusing the TRANS to VM button that is present on the SIP phones with our new speed dial button Direct VM Trans. Does anyone know how to...

Thank you. This is going to work perfectly for what we were trying to accomplish. Good thing I didn't trust our Avaya installers answer.

Hi Deploying some 9611G SIP phones in our lab and I've come across an interesting question. I'm wondering if it's possible to display the feature keys on the main screen along with the appearance buttons. I see you can do it easily for the H.323 phones but we are deploying SIP. Thanks for the help.

Wouldn't it be easier to just assign an ERL to the phone and program the ERL to an RLI that local terminates to a specific extension or to a route that has no members? We did this with a group of phones that had several misdials to 911 (even with misdial prevention turned on) added a new ERL...