Exchange UM integration to CUCM 4.x using IP-IP GW

A few days ago I printed the configuration for an IP-IP GW to integrate OCS with CUCM(all versions). I wanted to expand on this a little to include the configuration for Exchange UM to Cisco Unified Communications Manager 4.x(Callmanager). This is a very similar configuration and if you are using CUCM 4.x you could use the same IP-IP GW (Cisco router) to connect to the PSTN, Exchange and OCS, depending you load requirements of course. Depending on the dial plan configured in Exchange your configuration will vary for your dial peers. It is always best to match your extension length used in CUCM and the extension length in Exchange UM dial plan. The example below is based on 10 digits. Out bound dial peers to CUCM are required for the play on phone feature. This is a straightforward configuration in CUCM with the IP-IP GW configured as a H323 gateway. No MTP’s are required. Configuration unrelated to IP-IP GW has been removed for clarity. IOS version 12.4.15xz is required for this configuration.

!
voice service voip
no notify redirect ip2ip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service media-renegotiate
supplementary-service ringback h225-info
fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none
h323
emptycapability
no h225 timeout keepalive
h245 passthru tcsnonstd-passthru
sip
midcall-signaling passthru
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
!
!
!
!
dial-peer voice 103 voip
Description incoming dial peer for Exchange numbers
incoming called-number 555111110[12]
dtmf-relay h245-alphanumeric
codec g711ulaw
!
dial-peer voice 111 voip
description Dial-Peer To Exchange Subscriber Access
preference 1
destination-pattern 5551111100
session protocol sipv2
session target ipv4:X.X.X.X
session transport tcp
dtmf-relay rtp-nte
codec g711ulaw
fax rate disable
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 112 voip
description Dial-Peer To Exchange AA
preference 1
destination-pattern 5551111101
session protocol sipv2
session target ipv4:X.X.X.X
session transport tcp
dtmf-relay rtp-nte
codec g711ulaw
fax rate disable
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 113 voip
description to Callmanager 4.X FIRST SUBSCRIBER
preference 1
destination-pattern 1[2-9]..[2-9]......
session target ipv4:X.X.X.X
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 114 voip
description to Callmanager 4.X SECOND SUBSCRIBER
preference 2
destination-pattern 1[2-9]..[2-9]......
session target ipv4:X.X.X.X
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
!


Please post your comments and questions.

IP-IP GW Configuration Notes for OCS to CUCM Connectivity

Some time ago I wrote a document that Microsoft went on to release to a few companies for pilot. The document outlined the configuration for Cisco/Microsoft integration using the IP-IP GW concept. This is a reprint of some updated content here that should also hold true for OCS R2 and all versions of CUCM.
One of the important elements for this integration is the version of IOS for the Cisco router. Although no version of IOS is bug free, 12.415xz has proven to be stable and relatively bug free in most situations. The following IOS command line example is based upon 12.4.15xz which has features found in other versions but without some of the bugs from later version like 12.4.20t.



The configuration below gives some typical NANP dial peers. You may also want to include dial peers for international and 911. Depending on your normalization in OCS your dial peers will of course vary. One thing to note, is IOS (12.4.15xz)there is no need to specify the + sign for incoming or outgoing dial peers as this version automatically strip the +. This also means no need to modify the mediation server to remove the plus either. This example follows the diagram depicted with different version of CUCM with 5.x (this can also be 6.x) using SIP and 4.2.3 using H323. For the version using SIP (5.x) you will need to configure MTP resources in CUCM and mark them as required. For CUCM using H323 only the gateway configuration is required, MTP resources are not required. In follow up entries, I will talk more about CUCM configuration for using H323 or SIP in both versions.
!
voice service voip
no notify redirect ip2ip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service media-renegotiate
supplementary-service ringback h225-info
fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none
h323
emptycapability
no h225 timeout keepalive
h245 passthru tcsnonstd-passthru
sip
midcall-signaling passthru
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
!
!
voice translation-rule 103
rule 1 /^\([2-9].........\)/ /+1\1/
rule 2 /^\(...........\)/ /+\1/
rule 3 /^\(...........\)/ /+\1/
!
!
voice translation-profile AddPlusForOCS
translate calling 103
!
!
sccp local FastEthernet0/0
sccp ccm X.X.X.X identifier 2 version 4.1
sccp ccm X.X.X.X identifier 1 version 4.1
sccp
!
sccp ccm group 1
description SCCP CCM group
bind interface FastEthernet0/0
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 101 register MTP_5.x
!

!
dspfarm profile 101 mtp
codec g711ulaw
maximum sessions software 100
associate application SCCP

!
!
dial-peer voice 101 voip
description incoming SIP dialpeer
session protocol sipv2
session transport tcp
incoming called-number [2-9]..[2-9]......
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 555755 voip
description to OCS Mediation server
translation-profile outgoing AddPlusForOCS
destination-pattern 1555755[23678].
session protocol sipv2
session target ipv4:X.X.X.X
session transport tcp
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 111 voip
description to Callmanager 5.x or 6.x FIRST SUBSCRIBER
preference 1
destination-pattern 1[2-9]..[2-9]......
session protocol sipv2
session target ipv4:X.X.X.X
session transport tcp
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 112 voip
description to Callmanager 5.x or 6.x SECOND SUBSCRIBER
preference 2
destination-pattern 1[2-9]..[2-9]......
session protocol sipv2
session target ipv4:X.X.X.X
session transport tcp
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 113 voip
description to Callmanager 4.X FIRST SUBSCRIBER
preference 3
destination-pattern 1[2-9]..[2-9]......
session target ipv4:X.X.X.X
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 114 voip
description to Callmanager 4.X SECOND SUBSCRIBER
preference 4
destination-pattern 1[2-9]..[2-9]......
session target ipv4:X.X.X.X
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
!
!
gateway
timer receive-rtp 1200
!
sip-ua
timers connection aging 120
!

The IP-IP GW is a handy piece of software embedded in IOS code. The IP-IP GW can also serve the dual purpose of terminating T1/E1 along with the already mentioned functionality. It can also connect OCS to other PBX’s not capable of SIP. Of course this isn’t the only device capable of bridging the gap and now with direct SIP a small network would probably steer away from this idea altogether. Places where this architecture may suit are larger deployments where connectivity to multiple PBX’s is a requirement. The current limitation for a mediation server to connect only to one gateway is the main reason we stayed with the IP-IP GW. The required MTP resources required by CUCM can also prove somewhat of a pain unless your entire CUCM deployment is based on SIP. Using H323 can remove some of the configuration complexity when this is the only part of your CUCM deployment that is required to use SIP.

I will be posting follow-up CUCM configuration notes in the coming days, so stayed tuned.

Microsoft/Cisco Interoperability Choices

The unified communications sector is in a constant state of change. With each passing day, vendors seem to come out with new products. With these new or improved products come new interoperability options. When Microsoft’s Office Communications Server first arrived on the scene, interoperability with vendors like Cisco was limited but now with updates, OCS direct connect is viable option along with others. So what options are available for interoperability with Cisco Communications Manager (CUCM)?

SIP direct connect (OCS Mediation server -->SIP--> CUCM)

IP-IP Gateway (OCS Mediation Server -->SIP-->ISR or other device-->SIP/H323-->CUCM)

TDM Gateways back to back or TDM gateway to PBX (OCS Mediation Server-->SIP-->TDM Gateway-->T1/E1-->PBX)

Remote Call Control (OCS/LCS-->SIP/CSTA-->CUPS-->CTI-->CUCM)

There is a bunch of information on the web about direct SIP and RCC so for the next few blogs at least I will concentrate on using a Cisco ISR setup as an IP-IP GW(other vendors also have similar solutions). For large enterprises, this option can be attractive to allow a more scalable solution. It also removes compatibility issues and depending on the setup can allow a straightforward setup in CUCM. The IP-IP GW can enable OCS to CUCM (all versions) but also CUCM (4.2.3) to Exchange 2007 Unified Messaging.

Tip of the Day:

For those not familiar, the following TechNet forum offers an excellent source of information to supplement documentation and ask questions related to OCS telephony.

http://social.microsoft.com/forums/en-US/communicationsservertelephony/threads/

Welcome

Welcome to my first attempt at blogging on the WWW. Stay tuned in the coming weeks, months, etc as I share my experiences and thoughts on the Unified Communications industry and lesson's learned from working in a mixed Microsoft/Cisco VoIP environment.