SIp Line Requirement on CS1K

[telklass]

Hi Guys,

Avaya CS1K as one PBX for a Call Center infrastructure, AACC has been place to managed incoming calls and assign them to the desired skillset, other server are also part of the design to complete the entire Call Center infrastructure , such as SBC_Session Manager ect... We use Digital lines which are connected to the PRI card, so far everything works well. I only have one question for you Guys...

Are some of you already experienced the integration of a SIP line on Avaya CS1k ?

This what we are planing to do ,the replacement of our existing E1 line by the integration of the SIP line .

I only want to know the requirement .

Look forward to your reply.


Thanks !

 

[janaya]

Do you mean you are changing from an E1 to SIP trunks?

You will need :

CER - Customer Edge Router to accept the SIP trunks circuit. Normally an MPLS circuit from the carrier.
Internal switch - CER sends traffic through via a network switch
SBC - Session Boarder Controller - to move the SIP traffic from the Carrier to internal equipment
Session Manager - To route the call
CS1000 - you have

SIP trunks worked well for us, from a quality perspective. We are moving to AACC 7 SIP next year as well as Full SIP trunks from our carriers, so it should be interesting.

John Anaya
Amdocs Inc.
ACSS/ACIS - CS1000 Rls 7.5/Call Pilot 5
ACSS/ACIS - SME - IP Office 8.0
APSS/APDS - Avaya UC Services

Public Profile

http://www.linkedin.com/pub/john-anaya/25/1b9/a9b/

 

[telklass]

Yes,
THis is exactly what i plan to do, the existing E1 trunks will be replaced by SIP... Actually we already have SBC, from what i heard, short codes are created into it... and Session Manager is associated each short code then finally routed the call. AACC the one that managed all incoming before associated a call to a skillset. From what you said , is the router and only in the router , will be the first components to received the SIP circuit ? Obviously a SIP point will need to create on Session Manager. This is a full Call Center infrastructure and all the necessary hardwares are already in place. I guess you have a good knowledge since you did implemented this. Please posted any documentation that you think could help.

I look forward to your reply.


Thanks !

 

[telklass]

Hello,

Also which licence is necessary for a such configuration ,

i have printed out all license available i can see within the list : SIP ACCESS PORTS , AVAYA SIP LINES ,SIP CTI TR87 available and others , i want to know which license is required for a Sip line integration on CS1K.

Thanks !

 

[KCFLHRC]

Just to cut down on the confusion SIP Line in the Nortel world is actually SIP Phones. What you would need is SIP Access Ports. Here is a print out of one I maintain that has SIP Trunks for Aura Messaging as well as SIP Trunks for Experience Portal

REQ slt

System type is - Communication Server 1000E/CP PM
CP PM - Pentium M 1.4 GHz

IPMGs Registered: 5
IPMGs Unregistered: 0
IPMGs Configured/unregistered: 1


TRADITIONAL TELEPHONES 441 LEFT 38 USED 403
DECT USERS 0 LEFT 0 USED 0
IP USERS 5 LEFT 0 USED 5
BASIC IP USERS 0 LEFT 0 USED 0
TEMPORARY IP USERS 0 LEFT 0 USED 0
DECT VISITOR USER 0 LEFT 0 USED 0
ACD AGENTS 48 LEFT 19 USED 29
MOBILE EXTENSIONS 0 LEFT 0 USED 0
TELEPHONY SERVICES 0 LEFT 0 USED 0
CONVERGED MOBILE USERS 0 LEFT 0 USED 0
AVAYA SIP LINES 0 LEFT 0 USED 0
THIRD PARTY SIP LINES 0 LEFT 0 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 1 USED 0
SIP CONVERGED DESKTOPS 0 LEFT 0 USED 0
SIP CTI TR87 0 LEFT 0 USED 0
SIP ACCESS PORTS 20 LEFT 0 USED 20
RAN CON 35 LEFT 0 USED 35
MUS CON 10 LEFT 3 USED 7

IP RAN CON 0 LEFT 0 USED 0
IP MUS CON 0 LEFT 0 USED 0
IP MEDIA SESSIONS 0 LEFT 0 USED 0
TNS 65535 LEFT 64491 USED 1044
ACDN 24000 LEFT 23921 USED 79
AML 16 LEFT 15 USED 1
IDLE_SET_DISPLAY
LTID 65535 LEFT 65535 USED 0
RAN RTE 512 LEFT 510 USED 2
ATTENDANT CONSOLES 65535 LEFT 65534 USED 1
IP ATTENDANT CONSOLES 0 LEFT 0 USED 0
BRI DSL 10000 LEFT 10000 USED 0
DATA PORTS 65535 LEFT 65535 USED 0
PHANTOM PORTS 65535 LEFT 65529 USED 6
TRADITIONAL TRUNKS 65535 LEFT 65459 USED 76
ELC ACCESS PORTS 0 LEFT 0 USED 0
DCH 255 LEFT 251 USED 4

 

[PDTuser]

The supported method anytime SIP is routed on cs1k is to deploy ASM. This is addition to making sure cs1k is under control of System Manager as the PSS.
SIP access ports are required and the Node needs to have SIP GW application deployed. As stated SIPL (Sip line) is a different GW application only supported on the leader SS of a node, It is for registering SIP FW sets IE: (11xx/12xx w/SIP FW)which also requires licenses.
It is best to run the design by a certified Sales Engineer to make sure you have all the SW pkg and HW/licenses in your design.
CS1K=>NODE SIP GW=>ASM=>SBC=>carrier SIP

 

[telklass]

Thanks you guys, i am still on the state of writing the document as a plan work, then your advice guide me very much.

As a result all the necessary packages and licenses are available on the system, and Session Manger is the one server in place to route the SIP call (inbound / outbound ).

I beg you all that could comes with a suggestion, please post it right here .


Thanks !