CS1000 7.5 using Avaya Session Manager

[scsc78]

I have 2 CS1000 systems on Rls 7.5 tied together via Sip trunks using 6.2 System/Session Manager. When I place a call to site B from Site A using a CDP code of 88 plus extension, the phone rings, all good. But when I try the same type of call but the extension at site B is another CDP going to another CS1000 Option 11 using a PRI T-1 tie line I get reorder. I get a IP message 503 service unavailable.

I've tried everything in Sesion Manager to no avail. If I go back to using the PRI's between the sites, everything works fine.

Can I even do this?

 

[KCFLHRC]

So you are basically doing a trunk to trunk connection at site B, from a SIP trunk to a PRI circuit. Do you have media cards at site B to hand off the SIP call to the PRI ??

 

[scsc78]

Yes I have plenty of DSP's available. When i place a normal call to TDM phone it works fine. It's only when I do the trunk to trunk. The other location, the Option 11, is not a Sip End point. Only tied via PRI. I've looked at all restriction levels, NCOS, TGAR, NET data, etc. Everything looks normal.

If i go to another CDP that is a Sip Endpoint that works.

 

[KCFLHRC]

Ideally they would all be SIP end points, but I understand your dilemma. I have actually never tried tandem calls through a PBX that changes from SIP to PRI. You would think it would work. Maybe look at the settings in your SIP Route at site B and do a stare and compare on the old PRI route and see if anything sticks out.

 

[janaya]

Do you have the Global Plug In package. I believe there is a patch that has to be activated for cdp to cdp cals between sites. I think it may be Plug-in 4. It's built into the software but you have to turn it on.

To activate this you need the Global Plug In package. I don't believe there is a price when you order the upgrade, but if they don't put it in there you have to pay to have it activated. It is listed in the configurator as an additional package, much like DISA.

John Anaya
Signet6 Network Sciences
ACSS/ACIS - CS1000 Rls 7.5/Call Pilot 5
ACSS/ACIS - SME - IP Office 8.0
APSS/APDS - Avaya UC Services

Public Profile

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[allenmac]

Make sure that you are passing whatever digits are necessary to complete the call from the SIP endpoint system to the system connected via PRI tie.

So assuming your extensions are 4-digit, in the Adaptation Module for the destination endpoint you would enter the following in the Digit Conversion for Outgoing Calls from SM table:
Matching Pattern = 88 (Your CDP Access code)
Min = 6
Max = 6
Phone Context = (Your CDP domain)
Delete Digits = 2 (if you don't want to send 88xxxx, just xxxx)
Insert Digits = (leave blank, or enter whatever you would need to make the call reach the destination if this is a SIP to PRI tie call)
Address to modify = both
Notes = (Some type of description if you wish)

 

[scsc78]

thanks for the feedback, I'll try this tomorrow

 

[Voipstuff]

Login to the CLI of the SM and type – traceSM

Once up, select all 3 checkbox options to trace a SIP call. You can also filter the search by selecting "f" and entering -nf -no. After the trace, you will be able to isolate where the call is failing, then move into making the appropriate changes.

 

[scsc78]

Thanks for all the feed back. After trying everything that you guys said, I still couldn't get it to go. But during my testing I found another CDP number that went to a different Option 11 with the same type of setup. Those calls worked. So I went back to the basics and printed everything out. The route that did worked didn't have a TGAR and the one that didn't of course did. That's old school, I should have caught it.

But it's a learning curve with all this Avaya Red stuff working with the Blue. Thanks again.