Nortel BCM Interoperability SIP - Asterisk

Using the following setup, I have sucessfully added a link between a BCM400 and TRIXBox via SIP

Cavaet: No "security" ie. passwords etc has been tested - this is an "internal" configuration.

BCM400
IP Address: 192.168.100.253
ViOP GW Licesnes (licensed for SIP/H323)
A station to make/recieve calls on - I used the example of ext 3185

BCM Element manager:
Resources/Telephony Resources/IP Trunks
Routing table
Name: Trixbox
Destination Digits: 83
Destination IP: 192.168.100.252
GW Type: Other
GW protocol: None
VoIP Protocol: SIP

SIP Settings:
Domain name: - this will append to the SIP URI "unknown" name
Call Signalling Port: 5060
Outgoing Transport: UDP

URI Map:
Unknown/Unknown = 3100 (this will add to the domain name in call setup for the call context ie. call-context=3100@test.domain.com)

Telephony/Lines/Active VoIP Lines
Set all lines to be part of Line Pool "C"

Telephone/LinePools
Pool C: add DN 3185 (to allow 3185 to be allowed to use that pool)

Telephony/Dialing Plan/
Routing
Route: 083
External number: 83
Pool: C

Destination Code:83
Normal Route: 083
Absorb Length: all

That is all that needs to be set to allow VoIP Calls to be made or recieved

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Trix Box 2.2
Running on VMware-Server 1.0.3
Default installation
1x SIP phone connected via XLITE - ext8301
IP: 192.168.100.252

Add SIP Trunk:
Trunk Name: BCM400
Outbound Caller ID: 8300
Outgoing Dialing Rules: 31XX (dial this trunk for any call to extensions 31XX)
Outgoing Peer Details:

• host=192.168.100.253
• type=peer

Incomming Settings
User Context: from-trunk
User Details:
context=from-trunk
host=192.168.100.253
type=peer

Add an outbound route to send anything 31XX to BCM400 trunk

Add an inbound route to send anything from BCM400 trunk to extension 8301

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Once this is done you will be able to call from the Trixbox extensions to the Nortel extensions
CLID will display correctly.

• however*

making inbound calls from the BCM to the TRIX box will require you to allow all anonymous inbound connections
FreePBX/General Settings
Allow Anonymous Inbound Sip Calls = YES.

otherwise you will only get the IVR advising the number you have called is not in service.