[Asterisk-Users] Incoming SIP connection

[Joseph Rothstein]

Geetings to all.

I am having a hell of a time getting incoming SIP connections to work
properly, and am hoping that someone can help me. Here is what I am using as
a guide (from the wiki):

"Incoming SIP Connections

When Asterisk receives an incoming SIP call, the SIP Channel Module 
first tries to find a [user] section matching the caller name (From:
username), then tries to find a [peer] section matching the caller's IP
address. If no matching user or peer is found, the call is sent to the
context defined in the [general] section of sip.conf."

I am mainly concerned with the second point. I want to match an incoming SIP
connection to a particular IP address.

I have tried just about everything, and the connection always goes to the
default context, or the context defined at the top of the sip.conf file. I
would like to be able to direct incoming SIP connection to a particular set
of extensions. There is no username and password involved as there will be
many users coming from this one IP.

This is what I have tried recently:

[sipin_test]
type=peer
defaultip=195.27.242.120
context=test_trunk
deny=0.0.0.0/0.0.0.0
permit=195.27.242.120/255.255.255.255
dtmfmode=rfc2833
disallow=all
allow=ulaw
nat=no

I have also tried changing what is inside the brackets to the IP address. I
have tried many many different combinations of the above, but the IP address
never seems to get picked up correctly. 

I am testing the SIP connection using sipsak.

I realize that Asterisk is probably not the best SIP server to use, and plan
on migration to SER, but if anyone can offer any suggestions I would really
appreciate it.

Regards to all,
Joe