[asterisk-users] Sip trunk between
Asterisk and Mitel 3300 ICP
Joesph O asterisk at me.net.ng
Sun Jul 8 18:41:54 CDT 2007
Previous message: [asterisk-users] NAT
Next message: [asterisk-users] Asterisk and Mitel 3300 ICP
Messages sorted by: [ date ] [ thread ] [ subject ] [ author ]
hallo everyone,
fyi ... working SIP Trunk configuration between Asterisk and Mitel
3300 ICP
attached.
let's refine further, please test and share your feedback, regards,
Joseph Okoegwale
Abuja, Nigeria
-------------- next part -------------An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asteriskusers/attachments/20070709/4f7f15e1/attachment.htm
-------------- next part -------------SIP Trunk between Asterisk and Mitel 3300 ICP PBX
Source/Credit Timo Sariwating
http://www.sundancecommunications.com/forum/ultimatebb.php?/ubb/get_topic/f/6/t/000558.ht
ml#000000
Mitel 3300ICP = 192.168.1.2
Number range = 5xx
Trixbox = 192.168.1.101
Number range = 25xx
On the Mitel 3300ICP 1. Network Element Assignment
- create a network element for the local switch
- create a network element for each SIP peer, gateway, or Service
Provider
- create a network element for the Outbound Proxy if one exists in
your network
Ensure there is a local element for the Mitel. if there is none,
create it.
Create a network element for the Asterisk box
Name - Asterisk
Type = Other
FQDN or IP adress = 192.168.1.101 (Asterisk IP Address)
SIP Peer = selected
external FQDN or IP = 192.168.1.101 (Asterisk IP Address)
SIP registrar FQDN or IP = 192.168.1.101 (Asterisk IP Address)
Transport = UDP and Port = 5060 for all
2. System IP Port Assignment
Change the SIP UDP, TCP, or TLS port number if it is different from
the default value.
SIP UDP = 5060
SIP TCP = 5060
SIP TLS = 5061
3. DID Ranges for CPN Substitution
To set up the CPN Substitution table for outbound calls, enter a DID
number or a range of DID numbers assigned in the system.
Then enter the corresponding CPN substitution number that will be
delivered for that range e.g
Index = 10
DID Range = 500-599
CPN Substitution = 5XX
4. Create a SIP Peer for Asterisk
- Use SIP Peer Profile Form
SIP peer profile label = Asterisk
Local Account registration username = 150 (an extension that would be
used for authentication, should match in Asterisk)
Adress type = IP adress : 192.168.1.2 (ip address of 3300 ICP)
Authentication username = 150 (an extension that would be used for
authentication, should match in Asterisk)
password en confirm password = abcd (password set on extension, should
match in Asterisk)
Authentication = Challenge-based Authentication
Outgoing DID Ranges: select index 10 (select matching index if Calling
Party Number Substitution was configured)
5. Optional - SIP Peer Profile Assignment for Incoming DID
To associate a range of telephone numbers assigned by a SIP Service
Provider to a particular SIP Peer,
enter the required information in this form.
6. Trunk Service Assignment:
Configure the trunk as non-dial in or dial-in:
- update the Non-Dial-In Trunks Answer Point field for the incoming
calls.
- strip the number of leading digits in Dial-In Trunks Incoming Digit
Modification Absorb field
- add the appropriate number of digits in Dial-In Trunks Incoming
Digit Modification Insert field.
Trunk service number = 10 (based on my situation)
class of service = 64 (Enter the COS number that defines the required
options for the trunk)
class of restriction = 64 (Enter the COR number for the trunk. This
COR number must not have been assigned to a station (mandatory field))
trunk label = Asterisk trunk
Dial in Trunks Incoming Digit absorb = 0 (you can use this to do
leading digits absorption etc)
7. Class of Service Options Assignment
Enable the Public Network Access via DPNSS field in the class of
service for all devices that make outgoing calls through
SIP trunks, PRI trunks, LS trunks, and so forth that are connected to
SIP Trunks.
8. Route Assignment
Complete the following fields in this form:
- select SIP Trunk from the pull-down list in Routing Medium.
- select a SIP Peer Profile label from the SIP Peer Profile pull-down
list.
- enter a Class of Restriction group number in COR Group Number (this
determines which extensions *cannot* access this trunk, I am using a
COR that
permits all Mitel extensions to access the Sip trunk and therefore
call Asterisk users successfully)
- enter any required digits in Digits Before Outpulsing. (If this
field is left blank, digits will be sent out as "Enbloc".)
Route number = 10
Routing medium = SIP Trunk
Trunk group number = empty
SIP Peer profile = Asterisk
Route Type = PSTN access via DPNSS
- ARS Digits dialed Assignment:
Digits Dialed = 2 (2 is the first digit of my Asterisk extensions
Numbers to follow = 3 (5xx follow = 3 digits)
Termination Type = route
Termination number = 10 (route number created above)
Make sure to enable Public Network Access via DPNSS in the SIP trunk
COS.
On the Trixbox - Create a SIP trunk:
Outgoing Trunk name = Mitel
PEER Details:
allow=ulaw
auth=md5
context=from-pstn
host=192.168.1.2
insecure=very
nat=no
secret=abcd
type=peer
username=150
- Create a SIP Extension:
Display name = Mitel 3300ICP
Device options:
secret = abcd
canreinvite = no
context = from-internal
host = dynamic
type = peer
nat = no
port = 5060
dial = SIP/150
- Create an outbound route:
Route name = Mitel3300ICP
Dial patterns = 5XX
Trunk Sequence = 0 SIP/Mitel
- Create inbound routes for your SIP extensions:
for example SIP extention 2540:
- DID number = 2540
Set destination = core: extension 2540
Now you should be able to make call from SIP to Mitel and vice versa.
Previous message: [asterisk-users] NAT
Next message: [asterisk-users] Asterisk and Mitel 3300 ICP
Messages sorted by: [ date ] [ thread ] [ subject ] [ author ]
More information about the asterisk-users mailing list
