To Configure the Number Tab Settings:
1.On the Management Portal menu window, click System Configuration > SIP Soft Switch > Trunks/Gateways.
2.On the Trunks/Gateways page, click the Trunks/Gateways ID link, then click the Extension tab.
3.ID - The name of the trunk/gateway entry. Use this field as a registration username only when the gateway is configured for dynamic IP address registration with SIP Soft Switch.
4.Memo - The description of the entry the Trunks/Gateways page displays. Use this field to change the memo for the trunk/gateway after initial configuration, if necessary.
5.Trunk to conference server - Only select this check box for a Conference server trunk/gateway.
6.In the SIP Settings section:
•Static IP Address - 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 endpoints in your network can resolve it.
•Static SIP Port - The static SIP port of the trunk/gateway. If this field is blank, the trunk/gateway uses the default port number 5060. If the trunk/gateway uses a different port number, enter that port; otherwise, leave this field empty.
•Transport - Select the protocol for the endpoint to use to communicate with the SIP Soft Switch: TCP and UDP. UDP is the default.
•Keep-alive method - 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.
7.In the NAT traversal settings section:
•Registration expiration (Seconds) - If a SIP phone is behind the NAT router on a private network, while the SIP Soft Switch is on a public network or another NAT router, this option should be set to 30 (seconds) to ensure continuous data traffic between the SIP Soft Switch and the phone to keep the firewall/NAT mapping active.
•Use SIP Address from - Works in conjunction with the Registration expiration (Seconds) option, this option controls how the SIP Soft Switch recognizes the location of the phone: Default is used for non-NAT configurations; Remote uses the phone IP address as the host IP address from which the SIP REGISTER message was received; Contact uses the phone IP address from the Contact field of the SIP REGISTER message; Static uses the phone IP address from the Static IP Address field.
•Use SIP Port from - For phones behind NAT configurations, this option controls how the SIP Soft Switch recognizes the SIP port of the phone: Default is used for non-NAT configurations; Remote takes the phone's SIP port as the port from which the SIP REGISTER message was received; Contact takes the phone's SIP port from the Contact field of SIP REGISTER message; Static uses the phone's SIP port from the Static SIP Port field.
8.In the Media settings section:
•Remote Media Server - If using a remote media server, select that server from the list. For details, refer to Media Configuration.
•Bridge RTP - Select this option if you need audio to always route through Ivanti Voice (for example, if using a VoIP service provider).
•Preferred Order of Codecs for Calls to this Gateway - A comma-separated list of codec names (as defined by IETF RFCs) in order of preference (for example: PCMU, PCMA, GSM, G729). The SIP Soft Switch reorders the list of codecs when it propagates INVITE messages from phones to this trunk/gateway according to the specified order. This is useful for forcing endpoints to communicate using different codecs in different situations. For example, you can configure endpoints to use G.711 when communicating in a LAN and G.729 when communicating in a WAN.
9.In the Advanced settings section:
•Override domain part in Request-URI - Type the domain name or IP Address that will be used for the domain portion of the Request-URI (Uniform Resource Indentifier) 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 Ivanti Voice which would normally use the Ivanti Voice domain as the referring entity.
•Add P-Asserted-Identity: header to outgoing messages - Use this 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 response.
•CONTACT_USER - User in Contact header of outgoing request or To header of outgoing response.
•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.
If the specified macro value is an empty string, it will not be used to construct the header value.
Example: With sip:123%FROM_USER%[email protected] entered in the Add P-Asserted-Identity field and sip:[email protected] in the From header of an incoming SIP message, an outgoing SIP message will contain the P-Asserted-Identity: sip:[email protected] header.
10.In the Destination number/name prefixes section:
•Prefixes - The prefixes that access this trunk/gateway, as well as the cost of using this trunk/gateway for calls with that prefix. If a prefix can access multiple gateways/trunks with the same COR, the SIP Soft Switch compares the costs of each and uses the most inexpensive one to achieve Least-Cost Routing. Ivanti recommends using a relative value in the Cost field, as opposed to an actual monetary value.
Fill in the Prefix and Cost fields, then click Add to add another prefix.
11.Click the Update button on the tab to apply your changes.
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