How can I link mitel and trixbox? I have a mitel system on one site and a trix box on the other site. These two sites are on different networks interconnected by fiber. What hardware do I need to buy for both systems so that I can dial the extension from mitel to trixbox and vise versa. Please help
This is not an easy question to answer.
Linking a legacy system with an IP based system is one of the most difficult tasks you can undertake. You have to understand legacy interfaces and protocols and be ready to write custom dial plan code to make the Asterisk system look like it needs to for the Mitel.
You also have to know how the Mitel is configured internally to be compatable with the dial plan.
Depending on what Mitel (you did not provide much information) it could also connect via IP. More than likely you will use a Channelized T1 or PRI interface.
Many thnks for the responce actually the mitel system is a 3300CX with 50 IP phones connected to WS-3560V2 PoE switchand all working fine. I want to interconect this mitel to an asterisk system via IP just for internal dialing between 2 sites. These two sites are connected via fiber with 1841 routers on each side to seperate the 2 LANs.
Those phones are probably MiNet not SIP. Does the 3300CX have a SIP or h.323 gateway license?
The phone are actually mitel 5312 and the mitel 3300 have a SIP gateway license
Ok, if you have the SIP license simply build a trunk between the two machines. SIP is SIP, the keywords may be different but you only need a few parameters:
1 - Host (fixed IP no registration required)
2 - insecure=port,invite (tell FreePBX to trust this host)
3 - type = friend
4 - context = from-internal (allow access to the internal dial plan)
5 - disallow = all (turn off unneeded CODEC’s)
6 - allow = (list the codec you want to use)
7 - dtmfmode = (line this up with the MITEL)
These are the keywords that Asterisk can use. The FreePBX trunk tool frees you from using the Asterisk config files however the Asterisk documentation lists all SIP peer keywords and possible values.
Thanks, so in short you are saying that I dont need extra hardware?
No hardware required if the Mitel supports SIP and you have network connectivity between the Asterisk/FreePBX box and the Mitel.
It can be done with the SIP gateway license. I used to work at a place where we had several production 3300’s including a dev box in our lab. I was able to connect an Asterisk box to our dev 3300. It was about 2 years ago, so I wouldn’t be much help on what the steps are, but it has been tested successfully.
Tha sound interesting, are you able to tell me how you archieved that. Like how you configured the mitel and Asterisk box,what type of connection that you used and if at all they are any addionitiona hardware like routers etc.
dtbwalya, you are way over complicating the process. Building a trunk is a process of matching settings (on both systems).
I am not sure what the router question means. A router is a data networking device that connects two networks. If the FreePBX and the Mitel are not on the same network you will need a router. That is a network design question not PBX config related.
You start with the layer 3 connection, can the FreePBX ping the Mitel and the other way around?
Adam also told you you need a SIP Gateway license on the Mitel if you intend to connect to external SIP devices.
We’re doing what you want to do, and all we needed to purchase was a SIP TRUNK license for each connection we wanted to support between the two pbx.
SIP debugging in mitel is non-existent (as far as we can tell, and have been told), so if you run into any issues, wireshark is your friend!
we configured extension 1100 in asterisk as the mitel:
[mitel]
username=1100
secret=abcd
type=peer
allow=ulaw
auth=md5
host=192.168.0.2
insecure=port,invite
nat=no
we configured the mitel (important, or non-default values shown):
TRUNKS; SIP
SIP PEER PROFILE
SIP Peer Profile Label: Asterisk
registration user name 1100
maximum Simultaneous Calls 8 ;number of SIP trunks you purchased
SMDR tag 301 ;“T” plus this number is how SMDR shows calls to/from asterisk
User Name 1100
Password abcd ; must match asterisk info
Authentication option: Challenge-based
SDP Options (all NO except for):
Enable mitel Proprietary SDP YES
Signalling Options (all NO except for):
Disable Reliable Provisional Responses YES
we configured Outgoing DID Ranges: ;all our mitel extensions are 4xxx
DID range 4000-4999
CPN Substitution 4xxx
SIP Peer Profile Assignment by Incoming DID ; identifies which extensions exist on asterisk
Incoming range: 1100-1199
SIP Peer Profile Label: Asterisk