SIP Deployment in a multi-site environment

[7deedtz]

Hi All

We have an environment of 6 network regions, Avaya G650 servers on all Regions/sites. region 1 and 2 are the main sites housing Session Managers and System Managers. Currently have SIP Station and H.323 deployed on Region 1 and 2.Region 3 to 6 are on PRIs while Region 1 and 2 on SIP trunks. Each Region has its on DID/DDI range

We are trying to deploy SIP Extensions into the other regions. We can get the SIP extension to register with the SM on Either Region/Location 1 or 2 and can receive calls and dial other extensions however we cannot dial out.

When we run a trace we getting

08:50:48 SIP<SIP/2.0 404 Not Found (No route available)
08:50:48 Call-ID: 0e4ab5b547e419220545c9400
08:50:48 SIP>ACK sips:diallednumber@Domain SIP/2.0
08:50:48 Call-ID: 0e4ab5b547e419220545c9400
08:50:48 denial event 1166: Unassigned number D1=0x778b D2=0x201

LIST TRACE

time data
08:50:48 SIP>SIP/2.0 404 Not Found


Any ideas or suggestions will be greatly appreciated

 

[montyzummer]

Is that a true trace output ??

What settings do you have on your NR for your sip domains ,you need to have an adaptation in place in SM to replace the SIP>ACK sips:diallednumber@Domain SIP/2.0 to relevant details that the network (pstn recognizes), presently if the pstn is receiving SIP>ACK sips:diallednumber@Domain SIP/2.00 it does not know how to deal with that info and waves you off

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[montyzummer]

my guess will be you will need this adaptation
•
overrideSourceDomain (
may be abbreviated to
osrcd):
Adaptation module replaces the
domain in the From header (if administered), P-Asserted-Identity header and calling part of the
History-Info header with the given value for egress only.

have a look at this doc

http://downloads.avaya.com/css/P8/documents/100168166

page 267

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[kyle555]

Just a pointer that might be helpful - at a certain release (6 dot something... I can't remember...) CM and SM was able to look at the history info in a SIP message to associate a station to a network region. Perhaps for those NRs 3-6 you need to have the authoritative domain set in them.

Consider having your SMs in the core, and putting all your phones SIP. All your phones would signal through those SMs to CM to make a call out, and all incoming SIP traffic to CM would appear on your SIP SM trunk group belonging to a far-end network region in the signaling group.

What you'd ideally want is for that to be overridden by the IP in the history message of the phone making the invite such that CM may associate that station to a certain network region and location to place the call. That's available at the most current releases - maybe yours too.

The adaptation module mentioned in the post above would certainly do it, but being uptight as I am, I like to know why it didn't "just work" and I think some traceSM action and watching the invite leave CM for SM.

That all being said, if phones on sites 3-6 are on PRI, why would CM go back to SM? I would think your flow would be SIP phone-->SM-->CM originating app sequence-->treat digits to hit PRI--->leave PRI and CM would never necessarily be reaching back out to SM to complete that call unless you wanted the SIP phones at locations 3-6 to use the SIP trunks in the core.