Setting Up A Sip Server - Aastra Phone 312 User Manual

Setting up a SIP server

You can store the accounts of up to five different WLAN networks in your handset. The data required
for this is all stored as accounts in a list from which a server can be selected. The data for the SIP
account is defined by the system administrator.
E
Press Menu .
F
Select System and confirm using
F
Select Accounts and confirm using
F
<Active profile>
or
F
per DHCP
Press Options , select New
PIN ("22222" by default).
The empty entry mask for the account is opened to an SIP server.
If a network connection exists, the name of the gateway and the user ID (= call number) should
appear in the display within a few seconds of leaving the menu. If this is not the case, please see
"System > Accounts > Info" (see page 59).
"Info" will not be able to be called up if absolutely no access has been set up, and nothing has been
received via DHCP.
Parameter overview
Information regarding the menu control
In the setting windows of the profiles and access data, you can, by
move the cursor in the entry line by one position. By
the next entry line.
When making entries, you can determine the following accounts. Always switch between the
entry/selection lines using the arrow keys and finish the installation with
: Name of the System
System name
SIP server : The server that is to be used as an SIP proxy can be a hostname or an IP address. If a spe-
cial port number is required, it can be entered after a colon.
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OK .
OK .
and confirm with OK . Enter the administrator
shortly pressing the arrow keys,
keeping the arrow keys pressed , you can go to
OK .
Setting up a SIP server
Example: sip.aastra.com
sip.aastra.com:8200
172.30.203.12:8200
When making the entry, use the asterisk button to switch between different styles of writing: Let-
ters, numbers or IP addresses. In the "IP address" mode you can enter a dot using the pound button.
Registrar : If the registrar is not the same as the SIP proxy, its address can be entered here. If this field
is empty, the SIP proxy will also be used as the registrar. Therefore in normal configuration, this field
can be left empty (for entry format see SIP server example).
: An outbound proxy can be configured here. This can be necessary, e.g. if the hand-
Outbound proxy
set cannot solve the DNS names itself as a result of its configuration. The outbound proxy will be
entered as an IP address, or if necessary as an IP address:port. In most cases, this field can be left
empty.
Example : 172.30.203.12
172.30.203.12:6200
: This is the SIP user ID. This is normally the call number of this device. However, it is also pos-
User ID
sible to have a user ID that is not just made up of numbers.
Authentication name : Is used for authentications. May remain blank if the system does not require
an authentication or if the authentication name is the same as the user ID.
SIP password : This is used for authentications. Can be left empty if the system does not require any
authentication. This password is used independently of the inquired realm, as it is not necessary to
enter a realm.
: The voice codec, which should preferably be used. It influences the sound quality and
Pref. codec
also the bandwidth used on the network. If there is no particular reason to make a change, it is rec-
ommended to use the preset G.711.
DTMF method : It is possible to set for a SIP profile, how DTMF (MFV) will be transmitted.
Using the left softkey select . . .
(default setting): Transmits DTMF in the RTP stream according to RFC 2833 after
RFC 2833
the package type negotiated via SIP/SDP. If the package type is not negotiated, "Inband"
will be used automatically.
SIP-INFO : DTMF will be transmitted via SIP Info messages (no standard, however widely
accepted). This setting should be used if RFC 2833 is not supported.
RFC + INFO : Both transmission types are activated. Please note: Possibly, the other party
recognises numbers twice.
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