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VCS Integration with Avaya Communication Manager

Saurabh Gupta
Level 3
Level 3

Hi Experts,

Just a quick Question, Has anybody implemented integration of VCS with Avaya Communications Managers to allow Avaya VOIP Calls to Cisco Endpoints ( Audio Only).

Is it possible? If yes, How is it possible and what are the requirements and necessary configurations required? Only SIP/H323 Trunk?

Looking forward for your Valuable Inputs.

Warm Regards,

Saurabh Gupta

2 Replies 2

sveistef
Cisco Employee
Cisco Employee

Hi

I have seen an old, unofficial howto for how to connect a Nortel CS1000 to the VCS. Maybe that could help?

Configuring VCS

Configuration for VCS to work with Nortel CS1000:

This example has Nortel CS1000 with a SIP trunk to VCS.  The domain of the CS1000 is nortel-dplab.com

Calls to VCS from Nortel arrive with a prefix ‘58’ that is stripped.

  1. Set up a Neighbor zone
    • Enable SIP
    • Disable H.323
    • Set up IP address of Nortel in the Peer 1 address
    • Set advanced zone profile as shown (custom mode)
    • Set up a match pattern to route calls to ‘Number@nortel_domain’ …       allow an optional ‘;user=phone’
      - example shown below removes an optional prefix ‘654’ then passes       through 5digits@nortel-dplab.com provided the digits start with 234 or 56.  the ‘;user=phone’ may       optionally exist, and if it does will be passed through
    • Select Nortel Neighbor zone

  1. Set up 2 Transforms

  • Transform 1 converts E164 dialled H.323 numbers into a URI: numbers@nortel-dplab.com
  • Transform 2 converts the incoming number from Nortel ….
    - the received number is in a format which is not a URI but Number;phone-context= … @domain … etc. e.g. INVITE sip:58360;phone-context=cdp.udp@nortel-dplab.com;user=phone SIP/2.0
    - the transform removes the leading 58 (which I did not need for routing the call), takes the following digits then strips the remaining information off; it then appends the domain ‘@nortel-dplab.com’.

Most calls work, however there are a couple of known limitations and problems:

Limitations:

  • •1.  T150s must use L5.1 Release or later
  • •2.  Need to run VCS X4 or later

Problems:

  • •1.  SIP calls from VCS to Nortel Voice mail do not work; H.323 calls interworked to SIP do.
  • •2.  SIP devices directly registered to CS1000 register with a proxy-requires header.  CS1000 does not remove this so call to VCS fails

When it comes to a Deployment Guide for Nortel there are currently no official ones. This is an un-official working guide taken from the Forums (http://forums.na.tandberg.int/forums/showthread.php?t=3516&highlight=nortel+vcs&page=2 )

Rough working guide to CS1K SIP trunk (use at own risk!). This assumes the Nortel guy is present - DON'T TOUCH THE PHONE SYSTEM!

Tested with Nortel i2000 series phones and i2050 soft-phone.
Tested with TANDBERG MXP, Polycom VSX/Viewstation, Netmeeting, xMeeting

CS1000 Release 5.5
VCS X3.1

These steps assume you have a working CS1000 environment, with SIP Peer Networking running, ESN's for each CS1000 that trunks to VCS.

Nortel Config:

1 - Create a steering code (CDP steering code or UDP location code) that routes calls to the VCS.

2 - For UDP, create a CDP L0 domain for the TANDBERG VCS. For CDP, select the L0 domain the VCS will be a member of.

3 - Setup a Gateway Endpoint for the VCS call signalling IP address, configured as Static SIP.

4 - Create a routing pattern on the L0 domain that contains the VCS, and if INAC (access code) is required, build the route pattern accordingly.

Nortel Side Complete.


VCS Config:

1 - Create a neighbor zone on the VCS for the CS1000

2 - Turn SIP on, transport TCP, H.323 off and point to the CS1000 (or multiple CS1000's)

3 - Turn "Searches automatically responde to:" to "On"

4 - Turn "Empty INVITE allowed:" to "Off"

5 - Set your route pattern to begin with either the UDP access code (numeric) or the CDP steering code (numeric) you previously configured on the CS1000 for the VCS.

Note, if using Uniform Dialing Plan (UDP), you will need to add a phone-context to the outgoing dialed string. This is accomplished with a regex transform set to replace.

For example, if my INAC/CDP Steering code was '99', and the phone-context on the CS1000 side is "nortel@nortel.com", a zone match/xform would be:

Pattern String: ^99(\d+) (any number starting with 99)
Replace String: 99\1;phone-context=nortel@nortel.com


6 - Create a transform on the VCS to strip the "phone-context" field for inbound calls to the VCS from the CS1000 (assume the DN Prefix on the CS1000 routing entries field was '99' and the phone-context is "nortel@nortel.com")

Pattern: ^99(\d+);phone-context=nortel@nortel.com(.*)
Replace: \1

7 - Force the VCS to route all traffic for this zone, including Media, via the CLI:
"xConf Zones Zone (n) Neighbor SIP MediaRouting Mode: Latching

Note, Latching media results in a higer success of DTMF in my limited testing. Still, DTMF from CS1000 phones to MXP endpoints does not work. DTMF from MXP to CS1000 services does work however.

Your tining is perfect

We're in the process of implementing Avaya VoIP so the above should make life a lot easier for me.

Brilliant.

/jens

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