Search results for query: *

I figured so. Walking is good.

Is there a remote factory default command for UNISTIM Phones? I can factory default a UNISTIM or SIP phone using the **73639[MAC]## on the phone, and I can remote factory default a SIP phone by doing a SSH/PDT session with the phone and typing "reset2factory", etc, but I can't seem to do the...

Thanks for the offer, amriddle01. If I continue to get more headaches with this system, I may take you up on your offer.

I have a question on the Directory key for my 1230 SIP phone. I have a CS1000e release 7.5 with SIP phones on release 4. In my 1230DeviceConfig.dat file, I have my ADDR_BOOK_MODE set to NETWORK, and in my MAC file for this 1230 phone I define my address book file as 4010ADDR.txt. I am able to...

Sorry, wrong forum.

I have a question on the Directory key for my 1230 SIP phone. I have a CS1000e release 7.5 with SIP phones on release 4. In my 1230DeviceConfig.dat file, I have my ADDR_BOOK_MODE set to NETWORK, and in my MAC file for this 1230 phone I define my address book file as 4010ADDR.txt. I am able to...

The Avaya Support site has a link for all former Nortel Partners and Customer with KRS, ESPL, etc. login credentials to re-register with the Avaya Support website. It took me a day or two before I could get my login to the Avaya support website to be setup correctly. You may need to have your...

Has anyone implemented SIP or Unistim IP Phones on IPv6? We are thinking of implementing IPv6 here, and I'm just wondering how much more of an effort it is to configure it, especially when we plan to install an Avaya voicemail system. I do know that the Avaya voicemail system uses the Avaya...

The Authname would be the userID of the phone's TN. Here is what I used for my phone (I changed the domain name here) where I added two other phones userID to my phone's appearance. The documents state that the userID cannot be the same as the DN for KEY 0 in the PBX, so I put a prefix "AA"...

I finally got it to work using the alias in the provisioning script. I had part of the solution earlier but I missed a key statement - "PROMPT_ENABLE_YES" should be set to "YES". I followed the example in the NTP "SIP Software for Avaya 1100 Series IP Deskphones-Administration" under "Multi...

You'll need to purchase one "Third-Party SIP Line" license for each Cisco phone you will be installing. It costs more than a Nortel SIP license because they want you to buy Nortel SIP phones.

Nortel has a partnership with Unimax that has product called 2nd Nature. I saw a demo of it and 2nd Nature functions pretty much like the old Telephony Manager, which is not supported in release 7.0. There's not much documentation on it, except for Sales information.

Thanks. I'll check it out.

I've heard that 7.5 will be GA very shortly (end of November). I'll wait till then. Thanks.

I have a CS1000E with beta 7.5 software. I built a SIP phone on a 1220 telset and I was able to login and make and receive calls. When I tried to add an appearance of another DN to my phone (doesn't belong to anybody else), I cannot login the new number and get a 404 not found message. Is it...

I just came back from the test pbx and tried it again. In LD 97, I changed XCTP back to '15' and it immediately went back to the old "reaches wrong number" routine. Changing XCTP back to '14' was a little different in that the change did not work until I INI'd' the switch, then it worked. I...

Update - We are now able to dial out correctly. The problem was in the "XCTP" settings in LD 97, where we found the value to be '15' instead of the default '14'. We came across the solution in Nortel's "Knowledge Base" (old solution search) where a PBX wasn't sending out DTMF. The resolution...

Here's an interesting development. We rolled in a test PBX (opt 11 mini) running at release 4.5 and programmed the routes and trunks the same as our 5.5 (D4/AMI, TRK, DTN, etc, using a TMDI card) and our test PBX was able to call out correctly. Our PBX vendor is working hard on trying to...

LD 83 - REQ is LST --> this gives you a list of phones sorted in DES sequence: ADSC 15 OCT 2002 00 064 0 00 05 500 4D 65436 ADSC 15 OCT 2002 00 064 0 00 06 500 4D 65437 ADVTSG 16 AUG 2010 00 004 0 11 04 500 4D 68572 AGBICH 25 OCT 2002 00 048 0 11...

We actually split 2 T1 from the carrier (they said that we could) and moved one to our other PBX. Both PBX have SIGO at STD. Both PBX are programmed the same. The only difference is that the good PBX (CS1000M) uses DDP cards and the "not so good" CS1000E uses TMDI and a PRI gatewsy. The PRI...