asterisk users - Dec 2004 - Setting up asterisk for one user in private ip NAT.

Hi. 

I've just bought SIP telephony service from a Swedish telco.

I've managed to make and receive calls with kphone.

Now I want to set up asterisk to be able to add fancy features like
voice mail and recording conversations. But first I
have to get the basic setup right. I'm running asterisk and kphone on
the same machine, behind at NAT-router.

When I make a call (from my regular phone) to the SIP-number I get a
busy signal and I see my regular phone number in the debug output
of Asterisk. I guess that means I'm doing something right.

The problem now is that I can't get kphone or linphone to connect to
asterisk. Trying to connect from kphone to asterisk does not generate
any messages in the asterisk debug output. Non what so ever.

Which has me thinking that ip might be something with the
hostnames/ip-addresses that's not right? 

What does "bindaddr" do? I've tried changing it to my private IP
but
that doesn't make any difference.


I know that I'm not being very specific in my questions but I feel
that I need some handholding here. Some tests that I can run, for
example, to find out if my Asterisk setup is kosher. So, will someone
please hold my hand in this scary land of VOIP?

Alex

Here are my config files so far.

sip.conf
----------------
[general]
context=default               
port=5060                       
bindaddr=0.0.0.0
srvlookup=yes           

externip = <public ip of router>
localnet = 192.168.0.0/255.255.255.0         ; Internal NETWORK address
allow=ulaw
allow=alaw
allow=gsm
allow=all
nat=yes

register => xxxxxxxx:yyyyyyy@astrofix.rixtele.com/1000

[alex]
type=friend
host=dynamic
username=alex
secret=zzzzzzzz
context=outgoing


[rix]
type=peer
username=xxxxxxxxxx
fromuser=xxxxxxxxxx
secret=yyyyyyyy
host=astrofix.rixtele.com
fromdomain=astrofix.rixtele.com
context=sip-in
insecure=very
nat=yes
----------------------------



extensions.conf
----------------------------
[default]
exten => 1000,1,Dial(SIP/alex||t)


[sip-in]
exten => 1000,1,Dial(SIP/alex||t)

[outgoing]
exten => _0.,1,Dial(SIP/rix/${EXTEN}|20|t)


----------------------------

.qt/kphonerc
----------------------------
[Registration]
AutoRegister=No
SipServerSipUri="Alex Polite" <sip:alex@localhost>
UserName=alex
qValue
----------------------------

-- 
Alex Polite
http://polite.se

I've never tried softphones on Linux, but my guess is that since you run
kphone and asterisk on the same server you get a port conflict. If the
client uses port 5060 (default sip port) it would defenitely have problem
connecting to an asterisk on the same port.

Maybe you can change the kphone settings to use some other port or something
:)

/Anders 
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com 
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of 
> Alex Polite
> Sent: den 18 december 2004 14:53
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Setting up asterisk for one user in 
> private ip NAT.
> 
> Hi. 
> 
> I've just bought SIP telephony service from a Swedish telco.
> 
> I've managed to make and receive calls with kphone.
> 
> Now I want to set up asterisk to be able to add fancy 
> features like voice mail and recording conversations. But 
> first I have to get the basic setup right. I'm running 
> asterisk and kphone on the same machine, behind at NAT-router.
> 
> When I make a call (from my regular phone) to the SIP-number 
> I get a busy signal and I see my regular phone number in the 
> debug output of Asterisk. I guess that means I'm doing 
> something right.
> 
> The problem now is that I can't get kphone or linphone to 
> connect to asterisk. Trying to connect from kphone to 
> asterisk does not generate any messages in the asterisk debug 
> output. Non what so ever.
> 
> Which has me thinking that ip might be something with the 
> hostnames/ip-addresses that's not right? 
> 
> What does "bindaddr" do? I've tried changing it to my private
> IP but that doesn't make any difference.
> 
> 
> I know that I'm not being very specific in my questions but I 
> feel that I need some handholding here. Some tests that I can 
> run, for example, to find out if my Asterisk setup is kosher. 
> So, will someone please hold my hand in this scary land of VOIP?
> 
> Alex
> 
> Here are my config files so far.
> 
> sip.conf
> ----------------
> [general]
> context=default               
> port=5060                       
> bindaddr=0.0.0.0
> srvlookup=yes           
> 
> externip = <public ip of router>
> localnet = 192.168.0.0/255.255.255.0         ; Internal 
> NETWORK address
> allow=ulaw
> allow=alaw
> allow=gsm
> allow=all
> nat=yes
> 
> register => xxxxxxxx:yyyyyyy@astrofix.rixtele.com/1000
> 
> [alex]
> type=friend
> host=dynamic
> username=alex
> secret=zzzzzzzz
> context=outgoing
> 
> 
> [rix]
> type=peer
> username=xxxxxxxxxx
> fromuser=xxxxxxxxxx
> secret=yyyyyyyy
> host=astrofix.rixtele.com
> fromdomain=astrofix.rixtele.com
> context=sip-in
> insecure=very
> nat=yes
> ----------------------------
> 
> 
> 
> extensions.conf
> ----------------------------
> [default]
> exten => 1000,1,Dial(SIP/alex||t)
> 
> 
> [sip-in]
> exten => 1000,1,Dial(SIP/alex||t)
> 
> [outgoing]
> exten => _0.,1,Dial(SIP/rix/${EXTEN}|20|t)
> 
> 
> ----------------------------
> 
> .qt/kphonerc
> ----------------------------
> [Registration]
> AutoRegister=No
> SipServer> SipUri="Alex Polite" <sip:alex@localhost>
UserName=alex qValue>
> ----------------------------
> 
> --
> Alex Polite
> http://polite.se
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