3300 6 & CUCM 9/10 SIP Trunk Configuration Guide
Many thanks to moderator
martyn for allowing us to
re-print his excellent document.
This document is written as an update to the original configuration
guide from Cisco that covered the configuration of Mitel software
release 7.1 and CUCM 6.1. It is aimed at getting calls working
between the two systems and has not been tested for all functions
(such as MWI, conferencing, etc ) as the original document was done.
Further testing would be required to confirm interoperability on the
This document covers the configuration of a SIP trunk directly
between a 3300 and a Cisco CUCM system. Notes at the end of the
document cover the configuration of a Cisco Voice router rather than
CUCM as the terminating point.
This configuration is based on a MCD 6 system.
Earlier releases may contain different forms and could require extra
configuration. It is assumed that the system already has CoS and CoR
configured and that an understanding of the configuration of a Mitel
system is of an advanced level. No steps are given for where to find
forms appropriate to the configuration.
Under License and Option Selection check that SIP trunk licenses
have been applied to the ARID and have been allocated to the system.
new Network Element and include the IP address of the CUCM and set
the transport as default using port 5060. Make sure that the peer
is always active.
Trunk is crossing a lower bandwidth WAN link change the Zone from 1
to another number (that isnít the same as the controller) that is
unused on your system, to ensure that G.729 is preferred when SDP is
new Trunk attribute and assign a name, CoS and CoR appropriate to
the system. Ensure that a 0 (or appropriate number based on what
is being received from CUCM) is placed in the Dial in Trunks digit
mod absorb, as if left blank then calls will fail with a 503 error
in your SIP trace stating that the service is unavailable.
Note: if programmed incorrectly you will receive a SIP 503 error
new SIP Peer Profile. Assign a name to the profile, and select the
network element that you previously created. Set the Max calls based
on your license, and set the trunk service to the one created above.
the appropriate tabs as follows:
Options (Most Important Tab)
important options here are the Force sending SDP in initial invite
(SIP Early Offer), and prevent the user of 0.0.0.0. If Renegotiating
SDP is not enabled then calls will fail if both ends do not agree on
the correct codec up front, so this is important to enable.
Also ensure that the packetisation rate is the same on both ends,
otherwise calls will fail. Default is 20ms.
Signaling and Header manipulation
new entry in SIP Peer profile Assignment by incoming DID. Inlcude
the extension range of the mitel handsets, and select the profile.
Give the entry a name.
do not create this entry then you will end up with SIP 404 errors in
your SIP trace when calls are being made from CUCM to the Mitel.
If any outbound digit manipulation is required create an entry in
the DID ranges for CPN substitution and assign the entry under
the CUCM SIP Peer profile.
Create a new ARS route to point to the SIP trunk. Ensure that an
appropriate CoR is used, and digit
mod number is correct for the number of digits that may need to be
absorbed on outbound calls.
new ARS digits dialed entry(ies) for all ranges that are covered on
CUCM. Ensure that the correct termination number is entered based on
the ARS route created above.
configuration is based on a CUCM 9 or 10 system. Earlier releases
may contain different forms and could require extra configuration.
It is assumed that the system already has Device Pools,
Regions, partitions and CSSís configured and that an understanding
of the configuration of a Call Manager system is of an advanced
level. No steps are given for where to find forms appropriate to the
SIP Profile for use with the SIP Trunk. Suggestion is to copy the
pre-existing Standard SIP Profile. Assign a new name to the Copy
that can be identified as being for the Mitel trunk.
options can be left default for now. Save the new profile.
copy of the Non Secure SIP trunk Profile. Name the copy appropriate
to the use, and ensure that accept unsolicited notification and
accept replaces header are enabled
a new SIP Trunk.
Assign a name to the trunk. Select the appropriate device pool, and
MRGL. Call classification can be set as OnNet.
Inbound calls set the appropriate number of significant digits, as
well as a CSS for incoming calls. Assign a prefix if required.
Select Redirecting Diversion header delivery inbound to allow
features such as CFA to display the correct number when passing
across the trunk.
digit manipulation is required configure this under incoming party
settings. Generally this isnít required though for an internal
trunk between systems.
Under outbound calls select Redirecting Diversion header Oubound to
allow features such as CFA to display the correct number when
passing across the trunk. All other outbound settings can generally
be left default unless required specific to the environment.
information enter in the IP address of the Mitel system, and ensure
that port 5060 is the destination (this is the default). Select the
Security profile and SIP profile created earlier, as well as a
Rerouting CSS. Ensure that DTMF signalling is set to RFC 2833 for
inband DTMF signalling to work.
trunk configuration and reset the trunk for changes to apply.
now be able to receive incoming calls from the Mitel, if you have
configured your inbound CSS correctly to include the partition that
your phones are in.
For calls to be able to be made from CUCM registered phones to the
Mitel phones, there are multiple ways of configuring the routing,
but here we will use a route pattern.
Create a new RP, and assign the mask for the RP based on the
extensions on the Mitel. Assign a partition for the RP, and give it
Set the gateway to be the Mitel SIP trunk that you created above,
and ensure that the pattern is allowed to route. Call Classification
can be set as OnNet. Apply any Called/Calling party transformations
as appropriate to your environment.
RP. Remember that it will reset the trunk, so ensure that no calls
Assuming that you have configured your Partitions and CSSís
correctly you should now be able to make an outbound call to a Mitel
Mitel maintenance command line you can enable SIP tracing for
debugging of calls during the configuration process. To enable SIP
tracing enter SIP Trace On and when call problem has
been replicated you can turn it off with SIP Trace Off
Once completed the trace file can be downloaded via FTP from the /vmail
directory on the 3300. The file can then be opened with Wireshark
for analysis of the issue(s).
Click Here For the Mitel Forum
Related Forum Post
Please join our forum and follow our
newsletter. Your participation may help others.
found this article useful please