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Signalling server and SIP

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jensjuul

Programmer
Jul 28, 2007
50
DK
Hi,

I would like to setup a SIP trunk on my Meridian. Is that possible? Do you know where I may find a generel description of Nortel's SIP implementation and how it works?

Thank you very much.
 
Yes you need SIP access ports in your ISM license, and a sig server to start with.What RLS are you on.Did you check Nortel's site for some DOC's on it?
 
I'm sorry, but I am not too sure what RLS are - could you explain it to me?

I am able to connect to port 5060 on the signalling server, so that's a good sign :)

I've searched the documentation I received with the Meridian, but I cannot find anything. Do you know whether or not Nortel has a step-by-step guide to setting up a SIP Trunk?
 
lD 22 REQ Type ISS to get your rls, but I think that you might have some dificulty with this since you do not know how to even check RLS level... please don't take that personal.Also check in the same ld 22 at the REQ prompt type SLT to see it you have any SIP ports.
 
Hi again

Thank you so much - I've been reading a lot about SIP and I've just tried to do what you posted in your previous post. Here is the result.

REQ ISS

CALL SERVER/MAIN CAB
VERSION 2121
RELEASE 4
ISSUE 50 W +
IDLE_SET_DISPLAY NORTEL


REQ SLT
ANALOGUE TELEPHONES 128 LEFT 96 USED 32
CLASS TELEPHONES 0 LEFT 0 USED 0
DIGITAL TELEPHONES 80 LEFT 66 USED 14
DECT USERS 0 LEFT 0 USED 0
IP USERS 64 LEFT 2 USED 62
BASIC IP USERS 0 LEFT 0 USED 0
DECT VISITOR USER 0 LEFT 0 USED 0
ACD AGENTS 10 LEFT 10 USED 0

PCA 0 LEFT 0 USED 0
ITG ISDN TRUNKS 0 LEFT 0 USED 0
H.323 ACCESS PORTS 0 LEFT 0 USED 0
AST 1 LEFT 0 USED 1
SIP CONVERGED DESKTOPS 0 LEFT 0 USED 0
SIP CTI TR87 0 LEFT 0 USED 0
RAN CON 0 LEFT 0 USED 0
MUS CON 0 LEFT 0 USED 0
SURVIVABILITY 0 LEFT 0 USED 0

TNS 2500 LEFT 2239 USED 261
ACDN 300 LEFT 284 USED 16
AML 16 LEFT 15 USED 1
IDLE_SET_DISPLAY NORTEL
LTID 0 LEFT 0 USED 0
RAN RTE 500 LEFT 500 USED 0
ATTENDANT CONSOLES 2500 LEFT 2500 USED 0
BRI DSL 150 LEFT 150 USED 0
MPH DSL 0 LEFT 0 USED 0
DATA PORTS 2500 LEFT 2500 USED 0
PHANTOM PORTS 2500 LEFT 2469 USED 31
SIP ACCESS PORTS 6 LEFT 0 USED 6
TRADITIONAL TRUNKS 2500 LEFT 2470 USED 30
DCH 80 LEFT 78 USED 2
TMDI D-CHANNELS 64 LEFT 64 USED 0
KEY1 ########
KEY2 ########
KEY3 ########



Could you help me further? As far I can see, SIP isn't enabled? But why can I connect to 5060 TCP?

Thank you very much!
 
You have 64 Ip sets and use 62 that is why you got the signaling server for. But SIP or H.323 is not there. So you need to buy SIP licenses.
 
Ahh. OK. Thank you.

I've just tried to use a SIP phone. I've setup everything up in the NRS Manager, but I get a 501 SIP error as a result. So the server does not support REGISTER??
 
OK - thank you. I've just logged in to my Signalling Server.
Every 30 second I get this error:

31/08/2007 22:51:53 LOG0003 SIPNPM: CSTAAmlHandler_ItgMsgPBXAmlLinkRes: pbxAmlLi
nkInit( Unknown 0x1 )

Have you seen this?

The SIPCTIShow command returns this:

oam> SIPCTIShow
SIP CTI Status and Settings:
----------------------------
Application status: Initializing
Customer number: 0
Dialing plan: UDP
SIP URI format: dialstring
Maximum number of associations per DN: 3
-----------------------------------------------------------------------
CTI Dial Plan Information
-----------------------------------------------------------------------

Home Location Code: "Not Configured"
Country Code: "Not Configured"
NPA Prefix: "Not Configured"
INTL Prefix: "Not Configured"
LOC Prefix: "Not Configured"
SPN Prefix: "Not Configured"
NXX Prefix: "Not Configured"
-----------------------------------------------------------------------
CTI CLID Information
-----------------------------------------------------------------------

International Calls As National: TRUE
Subscriber / Number of digits to strip: 0
National / Number of digits to strip: 0
Area Code: "Not Configured"
Subscriber / Prefix to insert: "Not Configured"
National / Prefix to insert: "Not Configured"



 
SIP access ports are licenses for the SIP trunk you are looking for.
Go LD 21 LTM
you should find a rout defined as SIP (6 ports are already used)
 
SIP access ports are licenses? So I do not need to buy a license for the SIP CTI TR87 to make a SIP trunk available?
 
Not at all. As i can see in your licenses, there is already a SIP trunk defined. To mak it works, SIP parameters in the Node have to be defined. The remote site as well...
 
Then, what is "SIP CTI TR87"?

Isn't that when you need a SIP trunk? Fx to communicate with a softswitch using SIP?
 
CTI TR87 is for third party integration, such as integrating your PBX with Microsoft's LCS product.

 
That's what I want to do! I have developed a softswitch that communicates with the PBX via SIP. So in order to do that, I need to buy licenses, right?
 
FYI,

You'll need more than just the TR87 licenses. For every phone you want to control via CSTA, you'll have to have both a "SIP CTI TR87" license and a "AST" license. Having just the T87 license is not enough.

Given the slt output that you provided in an earlier post, you don't have any free ASTs, so you'll have to order those too.

In either case, these licenses aren't going to fix the "CSTAAmlHandler_ItgMsgPBXAmlLinkRes: pbxAmlLi
nkInit( Unknown 0x1 )" error that you're seeing. That comes out because you haven't configured the VAS correctly. If you do a stat on the ELAN that you've created you'll most likely see an application IP of 0.0.0.0. For example:

ELAN #: 032 DES: elan32
APPL_IP_ID: 0 .0 .0 .0 : 00000000 LYR7: DOWN DSBL

The key is the the SECU setting for the VAS you created. If you set the SECU to NO, then the AML will never come up. Set that to YES and re-enable your ELAN, and you'll be good to go (well... assuming you get your TR87 and AST licenses). They really ought to rename "SECU" to "would you like TR87 to work", that way, you can select the obvious choice of "YES", and not "NO". lol.

Cheers,
//Robert
 
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