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CS1000e SIP Trunking

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CSSMT

Programmer
Jun 23, 2010
20
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IN
Hi All,

We have a Nortel CS1000e installation. Now we need to integrate this with three sites running on Welltech SIP PBX 6200 and one site running on cisco call manager 4.1. somehow i have managed to bring all these four into my CS, i.e. the users of these PBX's are able to call my Nortel phones. But I am not able to make calls to any of these PBX.

This is what I have done so far.

1 Created D-Channel
2 Created 32 SIP Virtual Trunks
3 Created a SIP Route (route 1)
4 Created RLB (RLI 1)
5 Created CDP - DSC 60 (for cisco i am using 60)and FLEN 6 (as the dial plan is 4 digit for Cisco.
6 COnfigured NRS
a L0,L1, Gateway End point and Routing enteries (gave DN prefix 60)

please someone help and suggest what i am missing.

Regards
Goutam..
 
Goutam,

I haven't started testing this yet. Waiting on our SIP licenses to come in then I will be trying to try the CS1000E to a CM and to an Asterisk box. You have any tips for those of us that are a few steps behind you? Thanks!

Ryan
 
Hi Ryan,

you can download the Interoperability guide for Nortel CS with Cisco. that would give a good start.

regards
Goutam
 
CSSMT, I suggest you enable a SIPCallTrace via CLI in the Signaling Server and share what messages you receive during the call attempt from the CS1000 to the CM.

SIPCallTrace on
traceAllOn

If you don't see any messages then your CS1000 is blocking the call before sending it to the SIP Route in which case I will suggest you review the phone's restrictions.

Also is important to know what do you get on the phone when you are calling, fast busy? nothing?

 
Hi Arriero,

I will enable the trace, meanwhile when i dial the prefix, say 30 for the other location i get a busy tone and display on phone that says "Release and try again".

regards
 
That is a good indication that you have an issue with your NRS, are the Endpoints registered?
 
The IP Phones of Cisco and Welltech endpoints can call my Nortel. they are registered. But somehow i am not able to register to them.
 
Hi,

I have taken the trace of D Channel. this is what is coming.


FEAT :CRID
FEAT :CDS
FEAT :NCID
PROGRESS: ORIG ADDR IS NOT ISDN
CALLING #:9997 NUM PLAN: PRIVATE/ABBREVIATED (CDP)
CALLED #:1008 NUM PLAN: PRIVATE/ABBREVIATED (CDP)

DCH 1 IMSG CALLPROC REF 000001A0 CH 80 0 0 31 TOD 17:40:27

DCH 1 IMSG DISC REF 000001A0 CH 80 0 0 31 TOD 17:40:27
CAUSE :NO ROUTE TO DESTINATION

DCH 1 OMSG RELEASE REF 000001A0 CH 80 0 0 31 TOD 17:40:27

DCH 1 IMSG REL COMP REF 000001A0 CH 80 0 0 31 TOD 17:40:27



What route is it referring to?

below is the RDB

TYPE RDB
CUST 00
ROUT 1
DES SIP
TKTP TIE
NPID_TBL_NUM 0
ESN NO
RPA NO
CNVT NO
SAT NO
RCLS EXT
VTRK YES
ZONE 000
PCID SIP
CRID YES
NODE 100
DTRK NO
ISDN YES
MODE ISLD
DCH 1
IFC SL1
PNI 00001
NCNA YES
NCRD NO
FALT NO
CTYP CDP
INAC NO
ISAR NO
DAPC NO
MBXR NO
PTYP ATT
AUTO NO
DNIS NO
DCDR NO
ICOG IAO
SRCH LIN
TRMB YES
STEP
ACOD 8011
TCPP NO
TARG 01
CLEN 1
BILN NO
OABS
INST
ANTK
SIGO STD
STYP SDAT
MFC NO
ICIS YES
OGIS YES
PTUT 0
TIMR ICF 512
OGF 512
EOD 13952
DSI 34944
NRD 10112
DDL 70
ODT 4096
RGV 640
GTO 896
GTI 896
SFB 3


PAGE 002

NBS 2048
NBL 4096
TFD 0
EESD 1024
SST 5 0
DTD NO
SCDT NO
2 DT NO
NEDC ORG
FEDC ORG
CPDC NO
DLTN NO
HOLD 02 02 40
SEIZ 02 02
SVFL 02 02
DRNG NO
CDR NO
NATL YES
SSL
CFWR NO
IDOP NO
MUS NO
PANS YES
MANO NO
AUTH NO
TDET NO
TTBL 0
ATAN NO
OHTD NO
PLEV 2
OPR NO
ALRM NO
ART 0
PECL NO
DCTI 0
TIDY 8011 1
ATRR NO
TRRL NO
SGRP 0
ARDN NO
AACR NO

 
You have route 1 in Zone 000, normally zone 000 is the default zone and is configured with zone intent = MO.

Check zone configuration parameters in LD 117 or via Element Manager.

Create a new zone with zone intent = VTRK for your route 1 (SIP Route)
 
Hi Arriero,

I have added a new zone with Zine Intent-VTRK. but still the same problem.

any other suggestion?

regards
Goutam
 
Your original post says in the NRS config that you gave the DN entries prefix 60, but on the call trace the DNs start with 1 and 9.

From the d-channel message posted, I would think you need to add 10 as a routing entry to the cisco endpoint, as those are the digits you are trying to send.
 
I am using the DMI table to delete the prefix while dialing so the called number is coming as 1008 instead of 601008. mine is a four digit plan(9997 is my number).
 
I am a little confused....

If you are dialing 601008, then stripping the 60 using a DMI in LD 86, then the digits you are actually outpulsing are 1008.

Unless there is a routing entry that can resolve the digits 1008 against the cisco endpoint in the NRS, your call will not go through in this scenario.

For the call to work based on your dialing 60 as a prefix for the cisco site, then you must send 601008 to the virtual route, and then let the cisco strip the 60 off on it's inbound side.
 
Tried with that option also. But the result is same.

Anything to do with AC1 AC2?
 
not unless you are dialing AC1 or AC2.

If you do a SIP routing test in the NRS to simulate the call to the cisco endpoint, what are you using as the DN you are calling, and what is the result of the test?
 
i am using the DN as 60 and it says:

Route found
Routing type - regular route
gives the end point address, i.e cisco
sip protocol - TCP
sip port - 5060


DN type is : Private level 0 regional (CDP steering code)
 
That's good.

Does 601008 actually exist on the Cisco as a valid extension that can be dialled from other Cisco phones?
 
Goutam,

Did you assign the new zone (VTRK) to route 1?

Is Signaling Gateway configured as TCP for SIP?

When you dial 601008 at what point you get fast busy? Before or after you finish dialing all digits?


You never provided a printout for the SIPCallTrace I recommended. Can you post the results?
 
Hi Arriero,

Yes i have assigned the new zone to route 1
Yes the GW is configured as SIP
When I dial 601008 (Prefix 60, 1008 is cisco number) i get a fast busy after I finish dialing all the digits

OK I will take the SIPCallTrace and send post.
 
Hi All,

here is the SIP Trace:

07/07/2010 19:12:56 LOG0006 SIPNPM: SIPCallTrace: This is Outgoing Message
07/07/2010 19:12:56 LOG0006 SIPNPM: SIPCallTrace:
Message: Outgoing method INVITE(0) chid: 32 Called num: 2220 Far End Signaling IP: 10.0.32.12:5060 Transport:TCP CSeq: 1 INVITE
From: "Goutam"<sip:9997;phone-context=prestocdp.prestoudp.com@welltech.sipserver.az;user=phone>
07/07/2010 19:12:56 LOG0006 SIPNPM: SIPCallTrace:
To: <sip:2220;phone-context=prestocdp.prestoudp.com@welltech.sipserver.az;user=phone>
User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12
07/07/2010 19:12:56 LOG0006 SIPNPM: SIPCallTrace:
Media Info: 10.1.3.40 Codecs: G711 U-Law(0) G711 A-Law(8) G729(18) G723(4) Dynamic(101) Dynamic(111) Payload: 20 ms Media State: SIPNPM_MEDIA_SENDRECV

07/07/2010 19:12:56 LOG0006 SIPNPM: SIPCallTrace: This is Incoming Message
07/07/2010 19:12:56 LOG0006 SIPNPM: SIPCallTrace:
Message: Incoming response 404 Not Found chid: 32 Called num: 2220 Far End Signaling IP: 10.0.32.12:5060 Transport:TCP CSeq: 1 INVITE
07/07/2010 19:12:56 LOG0006 SIPNPM: SIPCallTrace:
From: "Goutam"<sip:9997;phone-context=prestocdp.prestoudp.com@welltech.sipserver.az;user=phone>
To: <sip:2220;phone-context=prestocdp.prestoudp.com@welltech.sipserver.az;user=phone>

07/07/2010 19:12:56 LOG0006 SIPNPM: SIPCallTrace: This is Outgoing Message
07/07/2010 19:12:56 LOG0006 SIPNPM: SIPCallTrace:
Message: Outgoing method ACK(1) chid: 32 Called num: 2220 Far End Signaling IP: 10.0.32.12:5060 Transport:TCP CSeq: 1 ACK
From: "Goutam"<sip:9997;phone-context=prestocdp.prestoudp.com@welltech.sipserver.az;user=phone>
07/07/2010 19:12:56 LOG0006 SIPNPM: SIPCallTrace:
To: <sip:2220;phone-context=prestocdp.prestoudp.com@welltech.sipserver.az;user=phone>
User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12



the SIP Trace gives the far end signalling server as 10.0.32.12, but this is my signalling server IP. the far end server ip is 10.1.2.56. thats possibly the reason why the entry of 2220 is not found in the server. can someone guide why it is showing my signalling server IP in place of far end.

regards:
Goutam
 
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