Hi,

I'm playing around with SIP trunking between our Mitel 3300ICP and an Asterisk (Trixbox) machine.

I have it working one way:

Asterisk -> Mitel is no problem
I can dial all Mitel ext, and even out over the ISDN BRI in the Mitel.

The other way around (Mitel -> Asterisk) is a problem though. Can't get the Mitel to use the Asterisk box as an Registrar.

Annyone else goofing around with this??

Mitel 3300ICP is running R7.0 UR2 and the Asterisk is a Trixbox machine ver 1.2.9.1

Using a Mitel 5220 in SIP mode and a Nokia E-70 connected to the Asterisk Box.

Any input on the subject is welcome.

------------------------------------------
Timo

Never mind. Have it sorted out already.

I had to delete the sip_proxy in the Mitel.
And remove the inbound rules from the SIP trunk in the Asterisk box.

Inspread I defined an ext. in the Asterisk box for the Mitel to use.

Hi,

I got an eMail with the request to post the programming step I did to use SIP trunking between te Mitel 3300ICP and the Trixbox. I thought that I would also post it here.

Mitel 3300ICP = 192.168.175.5
Number range = 2***
Trixbox = 192.168.174.200
Number range = 1**

Subnets are both local connected using a router.

On the Mitel 3300ICP:

-Network Element Assigment:
Create a network element for the Mitel and one for the Asterisk box
Asterisk=
Type = Other
FQDN or IP adress = 192.168.174.200
SIP Peer = selected
external FQDN or IP = 192.168.174.200
SIP registrar FQDN or IP = 192.168.174.200

- SIP IP Port Assignment:
SIP UDP = 5060
SIP TCP = 5060
SIP TLS = 5061

- DID Ranges for CPN Substitution:
Index = 10
DID Range = 2000-2999
CPN Substitution = 2XXX

- SIP Peer Profile:
SIP peer profile label = Asterisk
Local Account registration username = 150
Adress type = IP adress : 192.168.175.5
Authentication username = 150
password en confirm password = 150
Authentication = Challenge-based Authentication
Outgoing DID Ranges: select index 10

- Trunk Service Assigment:
Trunk service number = 10
class of service = 10
trunk label = SIP trunk
Dial in Trunks Incoming Digit absorb = 0

- Route Assignment:
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 = 1
Numbers to follow = 2
Termination Type = route
Termination number = 10

Make shure to enable Public Network Access via DPNSS in the SIP trunk COS.

On the Trixbox in FreePBX:

- Create a SIP trunk:
Outgoing Trunk name = Mitel
PEER Details:
allow=ulaw
auth=md5
context=from-pstn
host=192.168.175.5
insecure=very
nat=no
secret=abc123
type=peer
username=150

- Create a SIP Extension:
Display name = Mitel 3300ICP
Device options:
secret = abc123
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 = 2XXX
Trunk Sequence = 0 SIP/Mitel

- Create inbound routes for your SIP extensions:
for example SIP extention 100:
- DID number = 100
Set destination = core: extension 100

I didn't know how to use a wildcard in the inbound routes.

Now you should be able to make call from SIP to Mitel and viceversa.

Let me know if it was usefull to you.


Timo Sariwating

p.s. To get the Trixbox to dial out of the BRI line on the Mitel 3300ICP I also added another outgoing route:

Route name = ISDN
Dial patterns = 0XXXXXXXXXX
Trunk Sequence = 0 SIP/Mitel

This worked for me because most numbers in Holland are 10 digits and that was enough for testing.

Can this be done with a SX-200_ICP_3.1.0.11 MX??

No SIP on the 200ICP.

Thanks Timo,
Your guide helped immensely.My Mitel 3300 ICP extensions now talk to Asterisk extensions and vice versa.

Problem is that Asterisk extensions can not call out to the world through the E1 on 3300 ICP. I guess I need to configure something on the 3300ICP to permit Asterisk to route calls through it. Please help.

Here is an extract of the log on Asterisk whenever I try to call external world through 3300ICP, in this case, Extention 2540 on Asterisk called 92345678, 9 is a leading digit -

Jul 7 16:48:07 DEBUG[6860] chan_sip.c: Outgoing Call for 92345678
Jul 7 16:48:07 VERBOSE[6860] logger.c: -- Called Mitel3300ICP/92345678
Jul 7 16:48:07 DEBUG[6860] channel.c: Prodding channel 'SIP/2540-b7904a98'
Jul 7 16:48:07 VERBOSE[6860] logger.c: -- Started music on hold, class '24', on SIP/2540-b7904a98
Jul 7 16:48:07 DEBUG[6860] channel.c: Scheduling timer at 160 sample intervals
Jul 7 16:48:07 DEBUG[6270] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[email protected]' Request 102: Found
Jul 7 16:48:07 DEBUG[6270] chan_sip.c: Acked pending invite 102
Jul 7 16:48:07 DEBUG[6270] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Found
Jul 7 16:48:07 DEBUG[6860] channel.c: Generator got voice, switching to phase locked mode
Jul 7 16:48:07 DEBUG[6860] channel.c: Scheduling timer at 0 sample intervals
Jul 7 16:48:07 DEBUG[6270] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[email protected]' Request 103: Found
Jul 7 16:48:08 DEBUG[6270] chan_sip.c: Acked pending invite 103
Jul 7 16:48:08 DEBUG[6270] chan_sip.c: Stopping retransmission on '[email protected]' of Request 103: Match Found
Jul 7 16:48:08 WARNING[6270] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"Tester" < sip:[email protected] >;tag=as07fef065'
Jul 7 16:48:08 VERBOSE[6860] logger.c: -- SIP/Mitel3300ICP-0832de50 is circuit-busy

Hi, I am trying to do the same here, followed the instruction that you gave without success. When I call from the TB I get a recording "All Circuits are Busy." When I call from the Mitel I get Error display "Out Of Service" I am currently running Rle 7.1.6.10_1 on the Mitel.

Any thing else that I can do?

Hi, i am connected 3300ICP with asterisk using SIP trunking, but for some reason if i am calling from Mitel to Asterisk - everything fine, but if i am trying to call from asterisk to 3300ICP every time i am reaching main number(operator). I am not a working with Mitel a lot. What could be wrong? Any advices will be helpful. Thanks

Hi,
Well I am also connected with asterisk using SIP trunking, but I don't know I am unable to use Mitel. So anyone can suggest me that would be very helpful for me. Thanx
================
robert
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would anyone happen to know why i can get caller name and number from freepbx - mitel

but from mitel - freepbx i get Mitel3300 and 3300 as the number and not the number of the ext from the mitel


THanks

Ritztech, I'd suggest you start a new topic. Check the last dates. John C.

Timo, thank you so much for this... I'm attempting the same and this got me pretty far, but I'm still having same issue as your orginal problem - Aterisk to Mitel 3300 is fine, but Mitel to Asterisk. I'd REALLY like to resolve this because now I'm working with VoIP provider Vitelity.com and they host with Asterisk- so, I'm having the same issue with connecting to them (I was trying with internal Asterisk server before).


In order to make the promise of Open Source help save our small company $ from the "Mitel Mafia Cartel Racket Group" who charge crazy maintenance fees and nickle and dime and try to cheat those who do and don't know better- I need to get something else in place at least as a backup.

Has anyone else gotten this working who can send along some more details to the config?

Thank MUCHLY in advance!


P.S. - Vitelity.com has been great for a friend of mine, inexpensive and you can just setup a replensihment pool, gr8 for a backup system. Just which I could get my Mitel to talk to it!

Can anyone set this up for me?

I'll pay money.

PM me or email me directly: contact at jefferson kim dot com

Kim, Welcome to the board. Please start a new topic with your request. Also may need to be in the "Request and Installer" category, depending on what it is you specifically need. Let them know your set up.

The original topic is four years old. Locking this topic.

Thanks.