The Trunks/Gateways page
This section lets you configure each gateway and trunk in your network.
Each entry on the page is assigned a unique ID which includes a configurable list of prefixes the SIP Soft Switch uses to select the trunk or gateway to use for an outbound call.
Add Trunk/Gateway Entries
Create an entry in the dial plan for each gateway and trunk in your IPCM environment.
When adding trunk/gateway entries:
• Do not use the same ID twice.
• Configure the Number tab, Security tab, and Prefix Manipulation tab for each trunk/gateway
entry you add.
►To Add a Trunk/Gateway Entry:
1 In the Management Portal, click System Configuration>SIP Soft Switch>Trunks/Gateways). To
add an entry, use the empty fields at the bottom of the Trunks/Gateways page. Open the
online help topic "SIP Soft Switch Configuration: Gateway and Trunk Entries" for a complete
explanation of the fields and options.
2 In the ID field, type a unique alphanumeric ID for the trunk or gateway.
3 In the Prefixes field, type the number(s) to dial within the IPCM network to access the trunk/
4 In the Memo field, type a description of the trunk/gateway (optional).
5 Click the Add button. The page refreshes, displaying the entry in the table.
6 In the table, click the ID link for your new trunk/gateway.
7 Configure the settings on the Number tab. Click the online help icon for detailed field
a) The ID and Memo field settings carry over from the Trunks/Gateways page. Change the
settings if needed.
b) Only select the Trunk to conference server check box for a Conference server trunk/
c) In the Static IP Address field, type the static IP address or resolvable domain name of the
trunk/gateway. Change this field only if the trunk/gateway does not register with the SIP
Soft Switch. If using a domain name, make sure all extensions in your network can resolve it.
d) Type the static SIP port of the trunk/gateway in the Static SIP Port field. If this field is blank,
the trunk/gateway uses the default port number 5060.
e) In the Transport field, select the protocol for the extension to use to communicate with the
SIP Soft Switch: Default, TCP and UDP.
f) In the Keep‐alive method field, specify the SIP Method to use as a keep‐alive message in
mid‐call. For new trunk/gateway entries, the default is OPTIONS. Use INVITE if OPTIONS is
not supported by the specific SIP implementation.
g) If a SIP phone is behind the NAT router on a private network, but the SIP Soft Switch is on a
public network or another NAT router, set the Registration expiration (Seconds) field to 30
(seconds) to ensure continuous data traffic between the SIP Soft Switch and the phone to
keep the firewall/NAT mapping active.
h) The Use SIP Address from field works in conjunction with the Registration expiration
(Seconds) option, controlling how the SIP Soft Switch recognizes the location of the phone:
Default, Remote, Contact, and Static.
i) The Use SIP Port from field is for phones behind NAT configurations, controlling how the SIP
Soft Switch recognizes the SIP port of the phone. Options are Default, Remote, Contact, and
j) If using a remote media server, select that server from the Remote Media Server list. For
details, refer to the online help topic "Host Configuration: Remote Media Server
k) Select the Bridge RTP if you need audio to always route through IPCM (for example, if using
a VoIP service provider).
l) In the Preferred Order of Codecs for Calls to this Gateway field, type a comma‐separated
list of codec names (as defined by IETF RFCs) in order of preference.
m)In the Override domain part in Request‐URI field, type the domain name or IP Address that
will be used for the domain portion of the Request‐URI (Uniform Resource Identifier) of an
outgoing message. This setting should be used when the receiving gateway or trunk
requires the URI to be from a specific domain or IP address, and would reject transfers from
IPCM which would normally use the IPCM domain as the referring entity.
n) Use the Add P‐Asserted‐Identity: header to outgoing messages field to add a P‐Asserted‐
Identity header (per RFC 3325) to all outgoing calls (a common identifying header). Create
the header field value by utilizing a specified URI format (%macro% in the user section) with
any one of the following macros:
» FROM_USER ‐ User in From header of outgoing request or To header of outgoing
» CONTACT_USER ‐ User in Contact header of outgoing request or To header of
» FROM_DISPLAY_NAME ‐ Display name in From header of outgoing request or To
header of outgoing response.
» CONTACT_DISPLAY_NAME ‐ Display name in Contact header of outgoing request or
To header of outgoing response.
Example: With sip:123%FROM_USERfirstname.lastname@example.org entered in the Add
P‐Asserted‐Identity field and sip:John@frs.com in the From header, the header of an incoming SIP message, an outgoing SIP message will contain the P‐Asserted‐Identity: sip:123John456@frs.com header.
Note: If the specified macro value is an empty string, it will not be used to
construct the header value.
o) Use the Prefix and Cost fields in the Prefixes section to specify the prefixes that access this
trunk/gateway and the cost of using the trunk/gateway for calls with that prefix. The SIP
Soft Switch compares the costs of each and uses the most inexpensive one to achieve leastcost
routing. FrontRange recommends using a relative value in the Cost field, as opposed to
an actual monetary value. Click the Add button to add more Prefix/Cost pairs.
8. Click Update.
9 Configure the Security tab settings. Click the online help icon for field explanations.
a) In the Class of Restrictions field, select the class of restrictions you want to apply to this
b) Select the Trust this Gateway's IP Address check box if you want the SIP Soft Switch to
accept calls from this gateway without requesting a username and password. If unchecked,
the SIP Soft Switch requests the username and password you specify in the Username and
c) If the When trusting IP address, verify the message comes from static SIP port is
unchecked, a message can originate on any port of the host configured in the Static IP
Address field. If checked, a message must originate on the Static SIP Port of the host.
d) If you did not check the Trust this Gateway's IP Address check box, you must enter the
username and password this gateway uses to register with the SIP Soft Switch.
e) In the Enable Soft Switch to Register on this Trunk field, select Enable to allow the SIP Soft
Switch to register with a VoIP service provider on this trunk. If you select Enable, complete
the following fields:
» In the Remote Registration Number field, type the number the SIP Soft Switch uses
to register with the VoIP service provider on this trunk.
» In the Remote Registration Username and Password fields, type the username and
password the SIP Soft Switch uses to register with the VoIP service provider on this
» In the Remote Registration Interval field, type the number of seconds the SIP Soft
Switch waits before refreshing registration with the VoIP service provider on this
f) If outgoing registration is enabled, select the Replace ANI with Remote Registration
Number to enable the SIP Soft Switch to place calls to this trunk on behalf of the Remote
Registration Number instead of the actual originating party.
10 Click Update.
11 Configure the settings on the Prefix Manipulation tab. Click the online help icon for field
explanations and refer to the Prefix Manipulation Examples on page 48. If needed, click the
green + icon to configure multiple settings for the Destination or Origination Number.
12 When the configuration is complete, click Update.