Do I have a SIP phone licence?

[keefbeef]

I've printed off my SLT, but don't really understand some of the terms. I need to create a SIP extension. So I have any licences to allow this?


TRADITIONAL TELEPHONES 54 LEFT 28 USED 26
DECT USERS 0 LEFT 0 USED 0
IP USERS 2347 LEFT 754 USED 1593
BASIC IP USERS 636 LEFT 0 USED 636 OVERFLOW 160
TEMPORARY IP USERS 0 LEFT 0 USED 0
DECT VISITOR USER 0 LEFT 0 USED 0
ACD AGENTS 136 LEFT 21 USED 115
MOBILE EXTENSIONS 10 LEFT 10 USED 0
TELEPHONY SERVICES 250 LEFT 250 USED 0
CONVERGED MOBILE USERS 0 LEFT 0 USED 0
NORTEL SIP LINES 0 LEFT 0 USED 0
THIRD PARTY SIP LINES 0 LEFT 0 USED 0

PCA 175 LEFT 175 USED 0
ITG ISDN TRUNKS 0 LEFT 0 USED 0
H.323 ACCESS PORTS 172 LEFT 16 USED 156
AST 4375 LEFT 4330 USED 45
SIP CONVERGED DESKTOPS 250 LEFT 250 USED 0
SIP CTI TR87 400 LEFT 400 USED 0
SIP ACCESS PORTS 129 LEFT 109 USED 20
RAN CON 4 LEFT 0 USED 4
MUS CON 30 LEFT 0 USED 30

TNS 32760 LEFT 28242 USED 4518
ACDN 24000 LEFT 23931 USED 69
AML 16 LEFT 13 USED 3
LTID 32760 LEFT 32760 USED 0
RAN RTE 512 LEFT 511 USED 1
ATTENDANT CONSOLES 32760 LEFT 32760 USED 0
BRI DSL 10000 LEFT 10000 USED 0
DATA PORTS 32760 LEFT 32760 USED 0
PHANTOM PORTS 32760 LEFT 32588 USED 172
TRADITIONAL TRUNKS 32760 LEFT 32500 USED 260
DCH 254 LEFT 243 USED 11

 

[gwebster]

It is the "NORTEL SIP LINES" and "THIRD PARTY SIP LINES" which control the number of SIP phones you can program. Both of those are zero for you.

You have plenty of room to add UNIStim phones.

 

[keefbeef]

It's an ata adapter that I want to install. How about H323? Do I have any of those?

I'm thinking that these are H.323 ACCESS PORTS?

 

[bignose21]

Depending on the ATA just register it to the NRS put it in the NRS as a Static Endpoint and add the CDP routiong entries. Put DSC's in the CS1k to send numbers out to the NRS.

 

[bignose21]

No licences required for that BTW

 

[keefbeef]

I'm I right in saying that you don't need a DSC if you have VNR set to yes? It will go straight to NRS if it can't find the DN on the local system?

Just thinking about this. The DN will be on the local site, so you do need the DSC to make it look at NRS to find the ATA end point. How to you setup a DSC to push an entry to NRS. I'm use to pushing them to a routing list (RLI) entry.

Also, registering to NRS? - is that the same as a static endpoint?

 

[bignose21]

So in the past when I have used SIP ATA's you just set up a static SIP endpoint on the NRS and enter the IP address of the ATA (Some do support dynamic but it is probably easier for you to give the ATA a static address). Against the enpoint add the extension number(s) you are going to have on the ATA.

Yes like you say you can just use VNR or DSC both route to an RLI which sends them to the Virtual trunk DMI 0 as CDP.

To see if you got that bit right just enable the sip messaging on the lead sig server (type help sipnpm for commands (Turn on and set the level)) you should see the following:

SIP Invite to NRS
302 (Temp move message) from NRS
ACK Message to NRS
SIP Invite to ATA

Normaly then on the SIP ATA you just tie the number to the port (If it has multiple ports). Depending on the ATA you may need to set the register IP to the NODE address of the CS1000 for your outbound calls.

We mainly used to use Audiocodes which were fully compatible with nortel and could Dynamicaly register to the NRS. They had config for the endpoint name, NRS TLAN and you set the CS1000 in as the proxy for outbound.

 

[keefbeef]

Does the same generally apply for H323 trunks, onto a ATA ? Just a different protocol, but same setup?

 

[bignose21]

yep