# How do I configure Asterisk for sipgate?

Here you will find an overview of the settings for the basic configuration of Asterisk for sipgate.

{% hint style="info" %}
**Important:** Replace each **SIPID** with your **SIP ID** and **PASSWD** with your **SIP password**.
{% endhint %}

Please insert the following information into the **sip.conf** :

{% code title="sip.conf" %}

```ini
[general]
context=fromInternet
nat=yes
transport=udp
sipdebug=no
allowguest=no
alwaysauthreject=yes

register ➔ SIP-ID@sipgate.de:SIP-Password@sipgate.de/SIP-ID

[sipgate_incomming]
type=friend
context=sipgate_in
host=sipgate.de[sipgate_outgoing_account1]
type=friend
username=SIP-ID
fromuser=SIP-ID
fromdomain=sipgate.de
secret=SIP-Password
host=sipgate.de
nat=yes
insecure=invite
qualify=yes
dtmfmode=rfc2833
disallow=all
allow=ulaw,alaw
outboundproxy=sipgate.de
```

{% endcode %}

Please insert the following information into the **extensions.conf** :

{% code title="extensions.conf" %}

```ini
[fromInternet]
; this context should remain empty for security reasons

[sipgate_in]
exten => SIP-ID,1,Dial(SIP/extension) <-- instead of extension, you should define the corresponding peer
exten => SIP-ID,n,Hangup

[sipgate_out]
exten => _X.,1,Set(CALLERID(num)=SIP-ID)
exten => _X.,2,Dial(SIP/sipgate_outgoing_account1/${EXTEN},30,trg)

exten => _X.,3,Hangup
```

{% endcode %}

#### Switch the tone dialing method

To update the tone dialing method in Asterisk, change in the line **'dtmfmode'** the value from 'info' to **'rfc2833'**.


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