Using the integrated voice features in Communications Server 2007 R2 for external telephone calls adds efficiency and call controls when using Office Communicator. However, enabling Enterprise Voice within Communications Server 2007 R2 can be a daunting task. Using a SIP trunk by Broadvox and the following configuration steps makes the process easier and less costly.
Author: Brian R. Ricks
Original publication date: February 2010
Product version: Office Communications Server 2007 R2
There are multiple options available to configure PSTN connectivity to your Office Communications Server 2007 R2 infrastructure. The cost and complexity of integrating voice via Direct SIP varies depending on your phone provider, but a SIP trunk is a simple and cost effective solution. A SIP trunk configuration enables all Office Communications Server Enterprise Voice features, including simultaneous ring, call deflect, call forwarding, and the option to use routing groups. The SIP trunk configuration is simple with Broadvox.
Properly setting up a Broadvox SIP trunk with Communications Server 2007 R2 involves the following tasks:
Ordering the SIP Trunk from Broadvox
Supported SIP trunk providers can be found on the Microsoft Unified Communications Open Interoperability Program site at http://technet.microsoft.com/en-us/office/ocs/bb735838.aspx. Broadvox is a SIP provider I have worked with as well as other Microsoft MVPs to ensure interoperability with Communications Server 2007 R2.
Selecting a vendor that has not been qualified by Microsoft is a personal risk that you must take, but reach out to your local telephony providers and see if they offer a SIP trunk or give Broadvox a call at (800) 273-4320 (http://www.broadvox.com). The SIP trunk you order must support the Office Communications Server 2007 R2 implementation of Early Media Detection as described in RFC 3960 (http://www.ietf.org/rfc/rfc3960.txt), which has been an issue with the SIP providers I worked with (Broadvox corrected this issue with their latest Fusion platform) as well as TCP (rather than UDP) communications.
There are multiple Broadvox trunk offerings; I ordered the Go!Local plan, which offers unlimited local calls and long distance at $0.02/minute. Other plans exist that include unlimited long distance, 800 numbers, and even discounted international calls. From the website, you can complete a request for a call back form or you can call (800) 273-4320 to start the process. If you do not plan to port any phone numbers, the entire process is relatively quick, usually less than one week. After the order has been fulfilled, you are ready to configure the Mediation Server. When speaking to the sales representative make sure you specify the following:
Configure the External Firewall to Allow Communication Between Broadvox and Your Mediation Server
The settings you configure on your firewall will be unique to the firewall that you are using. Table 1 shows the ports and protocols on your firewall that are required for connectivity to Broadvox. Using the information in Table 1, create multiple firewall rules that allow inbound traffic from the Broadvox IP addresses for the selected protocol and ports to the gateway listening IP address of your Mediation Server. The gateway listening IP address is configured as shown in Figure 5 later in this article. Broadvox uses three redundant data centers across the United States and thus the multiple source IP addresses. If your firewall restricts outbound communications as well, you will need to allow the traffic to be two-way.
The rules in the firewall will apply to the external IP address assigned to the Mediation Server's network interface card (NIC) called the gateway listening IP address on the Mediation Server. Your firewall must support source network address translation (SNAT) to perform network address translation of the external IP address of your Mediation Server.
Table 1. Inbound/outbound communication for a Direct SIP trunk to Broadvox's Fusion environment
Inbound Traffic Type to Mediation Server
Broadvox IP Address
Create a Dial Plan to Route Outbound Calls to Your SIP Trunk
The creation of dial plans is simple and straight forward when using a SIP trunk. This section assumes that internal and external communications are functioning within Communications Server already. This article covers only the implementation of a SIP trunk into a working Communications Server environment, meaning that Communications Server is functioning internally and externally with the exception of Enterprise Voice.
Figure 1. Forest level voice properties
Figure 2. Enterprise Voice location profile
Figure 3. Phone normalization rule
Using the Enterprise Voice Route Helper application found in the Office Communications Server 2007 R2 Resource Kit will simplify the process of configuring and testing phone dial plans before applying them to Office Communications Server.
Figure 4. Voice route
Configure the Mediation Server to Use the SIP Trunk
To configure the Mediation Server to use the SIP trunk, you will configure the IP address that your internal Communications Servers use to communicate, and then configure the gateway listening IP address to the IP address that your SIP provider uses to communicate.
Figure 5. Mediation Server General tab configuration
Only valid properties will be listed in the drop-down fields, meaning that you will not be able to manually enter any of the information. Your server must have two IP addresses: one for internal communication and one for external communication, and the Edge server must already be set up and configured within Communications Server.
Figure 6. Mediation Server Next Hop Connections tab configuration
Enable Users for Enterprise Voice
While the Mediation Server is restarting, you can configure users for Enterprise Voice. The Direct Inward Dialing (DID) numbers provided by the SIP trunk provider must be directly associated to the user accounts' Line URI Enterprise Voice properties within Active Directory. The inbound communication will be received in the E.164 format so inbound call manipulation will not occur. Rather, a lookup for the phone number will automatically happen based on the line URI of the user account. If there is a match, the Mediation Server will route the call appropriately.
Figure 7. Voice properties of Enterprise Voice user
Configuring Communication Server 2007 R2 Mediation Server to work with Broadvox for PSTN calls is simple and straightforward. While the ordering process is not automated (you must contact or request contact from a sales representative), the entire experience is simple and painless.
To learn more, check out the following:
Communications Server Resources
We Want to Hear from You
This is great! I don't have Broadvox - I have Bandwidth.com SIP trunks. However, I'm sure with some creative troubleshooting, I can apply this for my next Skunk Works project. Thank you!
This is great! I don't have Broadvox - I have interoute.com SIP trunks. However, I'm sure with some creative troubleshooting, I can apply this for my next Skunk Works project. Thank you!
Agree with DW Hunter. This is extensible to any ITSP (with the caveat of early media detection support). Short, sweet, and to the point! :-)
Do you have a similar guide for Lync 2010?
David, not yet. If there is enough demand, perhaps Brian will write the next version. :-)
I'd like to see the same procedure for Lync 2010... one more vote!
I too would like to see the same setup info for Lync.