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Nortel BCM Interoperability SIP

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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: <set to something that describes your BCM ie. testbcm.domain.com> – 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

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

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.


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