SIP Trunking Test Results for Synapse SB67070 SIP Gateway from AT&T
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b. Enter the Account Name. The SIP account name appears on the Dial Plan
Settings page and the Trunk Reservation page.
c. For Max Calls, enter the number of simultaneous call sessions you purchased.
The maximum value is 16. Setting the Max Calls to a value that is less than
the current number of Trunk Reservations for the SIP Account will generate
an error.
d. Enter the Display Name. The Display Name is the text portion of the Caller ID
that is displayed for outgoing calls.
e. Enter the User Name as provided by XO. The User Name, also known as the
Account ID, is usually the company's main number. Synapse will only accept
digits for a User Name.
f. Leave the Auth User Name and Auth User Password fields blank.
5. Enter the SIP Gateway Account Registration Settings.
a. Select Static Registration.
b. Enter the number of seconds for Registration Expires. This setting applies to
dynamic registration. It is a re-registration timeout value sent to the SIP
Provider. This is usually overriden by a re-registration interval determined by
the service provider’s response. The default setting is 3600 seconds and
should only be changed on the advice of your service provider.
6. Enter the SIP Gateway Account Server Settings as provided by XO.
a. Enter the SIP Server Address or URL. Ensure that you do not enter any
spaces before or after the address or URL.
b. If necessary, enter the SIP Server Port. Port 5060, the default setting, is
typically used for SIP transmission.
c. Registrar Server Address or URL and Registrar Server Port should be left
blank.
d. Outbound Proxy Server Address or URL and Outbound Proxy Server Port
should be left blank.
7. Configure the Codec Configuration.
a. Enable or disable audio codecs. You can click Add > to add the selected codec
to the Enabled Codecs list, or click < Remove to add the selected codec to
the Disabled Codecs list. Note: XO does not support the G.711a codec.
b. Arrange the enabled audio codecs. Select a codec, then click or to
change the order.
The SIP Gateway uses the audio codecs in the order they are listed on a per
call basis. You can choose codecs based on the speed versus audio
performance required.
8. Click Apply to save your changes.
(15 pages)
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