Configuring a Siemens Gigaset phone for AAISP

Feature Notes
Supports 302 Redirect	No
Tested on FireBrick SIP Server	Yes
IPv6 Support	No

These notes are the same for configuring an S685IP, A580IP and N300A too.

N300 and Voiceless

click to view settings

On the web interface, go to:

Settings - Telephony - Edit 1, Show Advanced Settings

Gigaset N300 Settings
Setting	Value
Connection Name or Number	AAISP (Or whatever you like)
Authentication Name	Your Number
Authentication Password	Your Password
Username	Your Number
Domain	Can be left blank
Proxy Server Address	voiceless.aa.net.uk
Registration server	voiceless.aa.net.uk

Then you'll need to assign this account to a handset for incoming calls, this is set under

Settings - Telephony - Number Assignment

Don't forget to secure the base station with a new PIN!

Configuring the S685IP / A580IP / N300A

If there are problems in setting up the S685IP it is easiest to start from a factory reset condition. This can be achieved by holding the blue button on the front of the base unit while power cycling it. After this the handset will need to be re-registered with the base unit (hold the blue button on the base unit while power cycling the handset). The pin should be changed from its default value of 0000 using the handset.

Starting from a factory reset, the following items need to be set using the web server in the base unit.

IP address:

By default the base unit picks up an ip address using DHCP. This can be over-ridden by connecting to "Settings -> IP Configuration", selecting "Static" and filling in the address details as required. Then click on the "Set" button.

Audio settings:

In menu "Settings -> Telephony -> Audio" select "Voice Quality - Own Codec preference". For "IP1" remove all codecs except "G.711 a law". This setting will generally give the best audio quality.

Other settings:

If there are problems with recognition of dtmf tones by other systems consider setting "RFC2833" in the "Settings -> Telephony -> Advanced Settings" menu. Avoid using "Audio".

Check that the Number Plan is set to use the VoIP rather than the fixed (pstn) line.

If you're behind NAT, then you'll need Support to set your number to the old legacy 'Asterisk' server and use the server/proxy etc as a.speechless.aaisp.net.uk - Don't set the stun server though.

Also see VoIP Security