Connect three MXeIII through IP/XNET (No audio Mitel 3300)

[reval]

Hi Friends,
I have 3 MXeIII. A with Numbering 4XXX, B with numbering 7XXX, and C with numbering 2XXX
there is an IP/XNET between A and B and there is no problem with it, Also between A and C everythin is OK but
when 7XXX calls 2XXX, vice versa, called party rings but as off-hook there is no voice !!!!
what is wrong

I thing routing is true cuz B can calls C vice versa but I don't know why there is no speech !!!
Thank you in advance for your help

[ralph]

This is a networking problem not a Mitel problem.
When a call is set up between phones, the local controller is talking to the phone to tell it to ring.  Once the phone goes off hook the  controller drops out and the RDP packets bypass the PBX and route phone to phone.

So in your case, let's look at a sample call from 7000 to 2000:
7000 calls 2000
   7000 signals Controller B that is wants to make a call
   Controller B signals Controller A.
   Controller A signals Controller C
   Controller C signals 2000 to ring
   2000 signals Controller C that it has answered the call
      All controllers then drop out and the call runs directly between 7000 to 2000

So my conclusion is this:  the network that the phone 7000 is on cannot route packets to the network that controller C is on.

You can test this out by connecting a laptop on either the B network or the c network and pinging a 53xx phone on the other network.

Ralph

[bobcheese]

do your phones have the same gateway as their respective controllers? If firewalls between controllers have you opened required ports for just the controllers or for the entire voice subnet?

[shahriar]

HELP PLZ
Suppose we have two controllers
‘A’ = 192.168.180.20 &
‘B’ = 192.168.100.90 and
in between them there’s a VPN router (say V), so IP/XNET trunk is well established,
And calls/voice are being transferred properly among users (analog/DNI/multiline IP) of A & B.

*But if we register an IP phone (suppose 1234) with IP 192.168.183.x (different network) via router ‘R1’ to Controller A and
Another ip phone (5678) with ip 192.168.101.x via ‘R2’ to Controller B
1234 can call 5678; 5678 rings but no voice is being transferred or can’t hear anything.

but if add static route in R1 and R2 to reach 192.168.183.x network to 192.168.101.x each other then voice can be heard/transferred. That means the controller drops out and the RDP packets bypass the PABX and route phone (1234) to phone (5678).

*** But how voice can be routed through PABX without adding extra route on R1 & R2 routers, I mean voice should be routed via 3300 Controller like phone->System<->System->phone, (not phone<->phone)

[bobcheese]

@ Shahriar - Incorrect, IP RTP packets are peer-peer (phone to phone) PBX is only involved for call setup/event/teardown. There is no way to force calls to go via controller is communication is IP-IP

[JoeShmo]

we had this problem with our setup.  Turned out that we needed rules in a firewall for not only the MCD IPs but also the RTC and E2T IPs, so check those as well.  Also, you didn't indicate if you are dealing with IP phones or analog.  If IP, then alot of the comments above make sense; the two devices talk to eachother, but if analog to IP, or analog to analog, then the RTC/E2T get involved.

[shahriar]

@bobcheese:

so it can't be possible without static route in R1 and R2 to reach 192.168.183.x network to 192.168.101.x each other?

different departments are using 192.168.183.x and 192.168.101.x network, if i have to enable data passing between these 2 networks, then department privacy violets.

so i thouht that Controllers are getting each other so there should be voice to hear

[bobcheese]

You only need to open certain ports for RTP etc the route does not need to be 'all IP'

[mehrdad]

check the gateway of E2T

[martyn]

Do you have a Layer 3 network diagram that shows the connectivity between the sites?