[asterisk-users] SIP provider and NAT

[Jim Dickenson]

I have an Asterisk server, running 1.6.0, on my home network. It has an IP
address of 192.168.0.201. The server is behind my Linksys router that does
NAT for my home devices. I have a softphone on my Mac that is on the same
NATed network as the Asterisk server. It has an IP address of 192.168.0.1. I
have tried this with a Grandstream phone with the same results.

The Asterisk server registers with a SIP provider and seems to maintain a
registration.

When I try to dial out from the soft phone or the Grandstream via the SIP
provider I get a "Bad Request" response from the provider. I am guessing
that many of the packets have my internal address and need my public
address.

Is there some way to resolve this?

Here is the relevant entry in extensions.conf:

exten => _88X.,1,Dial(SIP/${EXTEN:2}@proxy01.sipphone.com,20,r)
exten => _88X.,n,Hangup()


Here are my entries in sip.conf:

[proxy01.sipphone.com]
type=peer
context=mine
disallow=all
allow=ulaw
dtmfmode=rfc2833
host=proxy01.sipphone.com
fromdomain=proxy01.sipphone.com
insecure=port,invite
qualify=yes
fromuser=myid
authuser=myid
defaultuser=myid
secret=mypwd
canreinvite=no

[GXP280]   
type=friend
context=mine
nat=no
canreinvite=no
host=dynamic
secret=mypwd
callerid=GXP280 <109>
mailbox=109 at ourvm
busylevel=2

[dickenson]
type=friend
context=empl
nat=no
canreinvite=no
host=dynamic
secret=mypwd
callerid=Jim Dickenson <108>
mailbox=108 at ourvm

Looking at the SIP debug transaction I do not see my public IP address at
all. There must be some setting about the server having at NATed address.

-- 
Jim Dickenson
mailto:dickenson at cfmc.com

CfMC
http://www.cfmc.com/


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