Installation guide

Dial Plan Configuration – SIP Endpoints
Acano solution: Single combined Acano server Deployment Guide R1.6 76-1054-01-H
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 Name e.g. Call 001
 URI e.g. 88001
Note: coSpaces can also be created from the API. See the API Reference guide.
5.2.4 Adding a dial plan rule on the Acano solution
1. Still in the Web Admin Interface, go to Configuration > Outbound Calls and add a dial plan
rule with the following details:
 Domain = vc.mycompany.com
 SIP Proxy = the IP address or FQDN of your VCS
 Local Contact Domain =
Note: The local contact domain field should be left blank unless setting up a trunk to
Lync (as in section 6.1.2).
 Local From Domain = acano.mycompany.com
 Trunk Type=Standard SIP.
SIP video endpoints can now dial into a call 88001 hosted on the Acano solution by dialing
[e-mail].
5.3 Media Encryption for SIP Calls
The Acano solution supports media encryption for SIP connections including Lync calls.
This is configured in the Configuration > Call settings page in the Web Admin Interface and
allows encryption to be Required, Allowed or Disabled for SIP calls made to or from the Acano
solution. Additionally, you can choose whether changes to this setting will apply to SIP calls
already in progress (Apply to Active Calls button) or just future calls by using the Submit
button at the end of the Call Settings page.
1. Sign in to the Web Admin Interface and go to Configuration > Call settings.
2. Select the appropriate SIP Media Encryption setting (Required, Allowed or Disabled).
3. Click either Submit or Apply to Active calls.
Note: The SIP Encryption field in the Web Admin Interface Configuration > Outbound Calls
page allows you to set the behavior for each Outbound Calls dial rule. This separates the
control and media encryption behavior, allowing a TLS control connection to be used in the
absence of media encryption, for example. (You can also control SIP control message behavior
via the API (see the API Reference guide.)
5.4 Enabling TIP Support
If you use endpoints such as the cisco CTS range, you require the new TIP protocol support
available in R1.6. Enable it as follows:
1. In the Web Admin Interface go to Configuration > Call Settings and in the SIP Settings
section, set TIP (Telepresence Interoperability Protocol) calls to Enabled.