[Asterisk-Users] 404 error found when making SIP point to point calls

[Guild Jackson]

Hi

I would like to make point to point SIP calls between
two digium with asterisk.
I have got the asterisk working with the digium fxs
interface but can´t get the SIP session working yet.
I have configured the sip.conf and extensions.conf
files the way that I can have calls between the
extensions 3002 (ip 10.5.0.4), located in one
asterisk, and 2008(10.5.0.2) located in the other one.
When I try to make calls between the two extensions,
from 3002 to 2008, for example, I got the following
message:

-- Got SIP response 404 "Not Found" back from 10.5.0.2

Bellow there´s the sip.conf and extensions.conf files
I have modified for the 2008 extension (the same way
I´ve done for the 3002 one):

======= sip.conf ==========

[general]
context=default
port=5060
localnet=10.5.0.2
maxexpirey=33600
defaultexpirey=33600
allow=g729
language=en
relaxdtmf=yes

localnet=0.0.0.0/0.0.0.0

[3002]
context=from-sip
type=friend
username=3002
fromuser=2008
host=10.5.0.4
allow=g729

[2008]
type=friend
context=2008
host=10.05.0.2
username=2008
disallow=all
allow=g729


======== extensions.conf =========

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp

[from-sip]
exten => 3002,1,Dial(SIP/3002,20,tr)

[default]
include => from-sip

What is wrong with these config files? Could anyone
help me?
What am I supposed to do or change on these files to
get the SIP working so I could make point to point
calls?
Does anyone have sip.conf and extensions.conf examples
to handle point to point calls?

Thanks in advance and best regards



		
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