How do I configure Asterisk for sipgate?

Configuration guide for Asterisk with sipgate – overview of sip.conf and extensions.conf settings.

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

Important: Replace each SIPID with your SIP ID and PASSWD with your SIP password.

Please insert the following information into the sip.conf :

sip.conf

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

register ➔ [email protected]:[email protected]/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

Please insert the following information into the extensions.conf :

extensions.conf

[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

Switch the tone dialing method

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