Mitel 3300 6 & CUCM 9/10 SIP trunk configuration guide 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 newer versions. 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.

Mitel 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.

System Configuration Under Licence and Option Selection check that SIP trunk licences have been applied to the ARID and have been allocated to the system.

Create a 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.

If the Trunk is crossing a lower bandwidth WAN link change the Zone from 1 to another number (that isn’t the same as the controller) that is unused on your system, to ensure that G.729 is preferred when SDP is negotiating.

Create a 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.

503 SIP error if configured incorrectly

Create a 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 licence, and set the trunk service to the one created above.

Configure the appropriate tabs as follows: Call Routing

Calling Line ID

SDP Options (Most important tab)

The most 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. Signalling and Header manipulation

Save the profile.

Call Routing Create a 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.

If you 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.

Create a new ARS digits dialled 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.

CUCM This 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 CSS’s 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 configuration.

System Configuration Create a 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.

All other options can be left default for now. Save the new profile. Create a 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

Create a new SIP Trunk. Assign a name to the trunk. Select the appropriate device pool, and MRGL. Call classification can be set as OnNet.

Under 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.

If any digit manipulation is required configure this under incoming party settings. Generally this isn’t 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.

In SIP 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.

Save the trunk configuration and reset the trunk for changes to apply.

Call Routing You should 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 a description. 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.

Save the RP. Remember that it will reset the trunk, so ensure that no calls are active. Assuming that you have configured your Partitions and CSS’s correctly you should now be able to make an outbound call to a Mitel extension.

SIP Traces From the 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).

Voice Router Configuration If you want to terminate the SIP trunk to a voice router, rather than CUCM then the following IOS configuration is required. Changes appropriate to the environment based on numbers used, and codecs allowed would be required. Additional dial peers for appropriate call legs are also required, but are omitted as they are environment specific. voice service voip address-hiding mode border-element allow-connections sip to sip no supplementary-service sip moved-temporarily no supplementary-service sip refer signaling forward unconditional fax protocol pass-through g711alaw sip header-passing midcall-signaling passthru ! voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 ! dial-peer voice 20 voip description *** Outgoing from SIP Router to Mitel *** destination-pattern 63.. session protocol sipv2 session target ipv4:x.x.x.x voice-class codec 1 no voice-class sip outbound-proxy dtmf-relay sip-notify no vad ! ! dial-peer voice 40 voip description *** incoming from MITEL to RTR**** session protocol sipv2 session target ipv4:x.x.x.x incoming called-number 72xx voice-class codec 1 dtmf-relay sip-notify