Aastra M720, Руководство по установке

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AMT/PTD/TR/0001/3/7/EN Installation Manual for digital sets - IP sets - SIP sets
Page 10-28 05/2008 Aastra Phone 312
10.7.1 Overview of the parameters
Remark on menu control
In the profile configuration and access data windows, press the arrow keys briefly to move up the
cursor one in a field. Press the keys longer to access the following field.
During input, you can determine the following accounts. Always change from input lines to
selection lines using the arrow keys then finish the profile configuration by pressing
.
System name : System name
SIP server : The server to be used as SIP proxy may be a hostname or IP address. If a particular
port number is necessary, you can indicate it after a colon.
Example: sip.aastra.com
sip.aastra.com:8200
172.30.203.12:8200
While making the entry, you can use the star key to switch between the different writings: letters,
figures or IP addresses. In IP address mode, you can enter a point using the hash key.
Registrar : if the registrar is different from the SIP proxy, you can indicate its address here. If this
field is empty, the SIP proxy is also used as registrar. This field may also remain empty in standard
configurations (to know the input format, see the example of the SIP server).
Outbound call : you can configure an outbound proxy here. It may be useful, for example, when
the handset itself cannot resolve the DNS names due to its configuration. Indicate the outbound
proxy as IP address or, possibly, as IP:port address. In most cases, this field may remain empty.
Example: 172.30.203.12
172.30.203.12:6200
User ID : this is the SIP user ID. Generally, it corresponds to the phone number of this device.
Nevertheless, it is possible to exclude any figure from the user ID.
Authentication name : It is used for authentication. It may not be entered if the system does not
require any authentication or if the authentication name is identical to the user ID.
SIP password : it is used for authentication. It may remain blank when the system does not require
any authentication. The password is used independently of the required realm (authentication
domain); so, it is not necessary to indicate this latter.
Preferred codec : This is the preferential voice codec to be used. It has no impact on voice quality
or on the bandwidth used on the network. When there is no particular reason to modify it, the
default codec G.711 is recommended.
DTMF method : For an SIP profile, you can determine how the DTMF (MFV) tone must be
transmitted. Select any of the following options using the left function key ...
RFC 2833 (presetting) : transmits DTMF tones in the RTP flow in accordance with
standard RFC 2833, after negotiating the packet type via SIP/SDP. If there is no
negotiation of packet type, the Intraband type is used automatically.
SIP INFO : The DTMF tones are transmitted via SIP-INFO messages. (They are not yet a
standard, but they are very widespread.) Use this option when standard RFC 2833 is not
supported.
RFC + INFO : Both transmission modes are active. Note that in some cases, there may
be some duplication in the digits detected on the correspondent’s side.
Intra-band : Like for voice messages, DTMF tones are transmitted in form of RTP
packets. Generally, only Codec G711 can be used to detect them. Select this option only
if the correspondent does not support any other mechanism.