VoIP Phones - Gigaset N300
|Supports 302 Redirect||No|
|Tested on FireBrick SIP Server||Yes|
These notes are the same for configuring an S685IP, A580IP and N300A too.
Most probably, all outbound traffic will already be enabled, so it is just necessary to permit essential incoming traffic to the S685IP. The following inbound settings are needed:
For SIP control data, allow UDP ports 5060 from 126.96.36.199-119
For RTP audio, allow UDP ports 5004-5020 from anywhere.
Use a good password and consider using IP lockdown on the AAISP control pages to help prevent fraudulent use!
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.
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.
Registration with AAISP:
Select menu "Settings -> Telephony -> Connections" and click the "Edit" button for IP1. Then click the "Show Advanced Settings" button and fill in the following fields:
Authentication Name: <aaisp phone number> (this can be in either UK or international format, for example +44203095nnnn or 0203095nnnn )
Authentication password: <password> (This is the same password that is set on the AAISP control pages for this number. Some non-alphanumeric characters can cause registration failure. In particular avoid "#" and "/")
Username: <aaisp phone number>
Domain: proxy.aasip.co.uk (Don't accidentally type aaisp!)
Proxy server address: proxy.aasip.co.uk
Registrar server: registrar.aasip.co.uk
All the other fields can be left as they are unless you are using NAT in which case the "Stun enabled" button should be selected and the "Stun server" set to stun.aaisp.net
When everything is entered click on the "Set" button. The display will return to the "Settings -> Telephony -> Connections" menu. Check that "IP1" is active and that the status is "Registered". You may wish to disable "Gigaset.net" in this menu. It should now be possible to make and receive calls.
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.
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