Here is a text doc I wrote for another guy I work with. It should be accurate but I wasn't in front of a BCM when I wrote it. Hope this help! There is a better way to setup the dialing plan than what I wrote but it will work.
2 BCM 400 4.0
SIP lines
MCDN keycodes on both sides
Site A (Host side) has PRI already setup and will be used to call out from remote site B.
Example using 4 digit DN's
If, when performing the routing and destination codes the system gives an error, it may be due to the DN's of the phone system interfering with the dialing plan. If this is the case the DN's wheather acive or inactive must be changed before that destination can be used. This applies to Dn's that are on a physical port or application port.
example: Cannot add destination code of 3 for VoIP trunks.
Look at active, inactive, or application DN's for DNs starting with 3, and change them to anything else but 3.
To avoid problems with multiple sites use a 4 digit plan. Destination code can be 32 for example (32xx DN's) and will work if the 2 does not interfere with dialing plan on local BCM.
1. Telephony, Lines, active VoIP lines. Click on each line and add it to the same pool group such as "A" and this pool should be assigned to any sets which will be calling across the VoIP trunks to remote site. Telephony, Sets, Active Sets, click on DN, line access, then Line pool access, add "A"
2. If seperate dialing plans example: DN 2xxx one side A, and DN 3xxx on side B
3. Must use CDP with same private network ID on both sides, location code unused, and private DN length
4. MCDN settings check TRO or else the VMail will not hear your voice when leaving a message on remote site.
TAT can be checked but not necessary with only 1 remote site.
5. IP routing: Resources, Telephony resources, then click on IP trunks, then at bottom the details should show up. Add new route with first digit of destination, name description, IP address of remote side, GW type is BCM, GW protocol SL1, VoIP Protocol is SIP, and QOS monitor and TX Threshhold can be left untouched.
Example:
Name= To BCM remote
Destination digits= 3 (for 3xxx digits)
Destination IP= 192.168.10.50
GW Type= BCM
GW Protocol= SL1
VoIP= SIP
6. Telephony, Dialing plan, Routing heading. Add new route, use pool that was created for VoIP lines earlier, DN type private.
7. Destination codes tab. Add destination code. Example: 3 (for 3xxx extensions), normal route is route just previously created which is probably 002, absorbed length has to be 0, no wildcards.
8. Telephony, Lines, Target lines. must add private received digits to allow calls between the 2 sites. Example: Site A extention 2100 needs private received digits of 2100. Appear&ring or Ring only under assigned DN's tab. Prime set and Control set should be 2100.
9. If DID's are used and routed to Remote site B, must have received # from telco at site A where PRi is, then redirect line to 3xxx set.
Site B Remote side
1. Find DN of Voicemail on Site A, Feature 985 so it can be added to forward to VMail.
2. Follow steps above up step 7.
3. To dial the public network through VoIP to site A's PRI add public route, Pool A, destination code 9, absorb length 0. Under Public network, add 1 digit to DN length for default, the add new DN prefixes with a 9 in front and the length should be 1 digit longer as well. This is because Site A hosting the PRI sees the VoIP trunks as remote dial-in from the private network and adding the 9 tells the system to grab an open line from the PRI pool to dial out.
Example:
DN prefix= default DN length= 11 (normally this would be 10 xxx-xxx-xxxx)
DN prefix= 9911 dn length= 4 (normally this would be 3 if 911)
etc.....for 411 9411, 1 91
4. for voicemail to work must go to Applications, voice messaging/contact center. Center 1, external number is DN of voicemail of Site A