[Asterisk-Users] Asterisk, IAX and NAT issue

Dan dtoma at fx.ro
Thu Jun 26 03:38:42 MST 2003


Hi,

I have two Asterisks identically installed on two computers.
One of them is directly connected to the Internet, the other one through a
NAT router (Netgear MR314).
On the one behind the router I have an X100P card installed for PSTN
connections.
In the local LAN of each PBX they are several hardware IP phones (Cisco 7960
and 7940 with SIP images, firmware image P0S3-04-4-00.bin).

I have defined the following extension in the one behind the NAT:

exten => _3XX,1,Dial(IAX/apbx at x.y.z.u/${EXTEN})
exten => _3XX,2,Hangup
exten => _3XX,102,Hangup

where x.y.z.u is the IP address of the PBX directly connected to the
internet.
apbx is the IAX user defined on that PBX in iax.conf file like that:

[apbx]
type=user
username=apbx
auth=plaintext
permit=n.n.n.n/255.255.255.0        ; n.n.n.n external NAT router address
host=dynamic
context=fullaccess
;allow=all

In this way, I can call any 3XX extension using an IAX connection between
the two PBX's.
It works perfect.

Now, on the other PBX I have the following in extensions.conf

exten => _1XX,1,Dial(IAX/pspbx at n.n.n.n/${EXTEN})
exten => _1XX,2,Hangup
exten => _1XX,102,Hangup

where n.n.n.n is the external IP address of the NAT router.
pspbx is the IAX user defined on the PBX behind the router, in iax.conf file
like that:

[pspbx]
type=user
username=pspbx
auth=plaintext
permit=0.0.0.0/0.0.0.0
host=dynamic
context=fullaccess
;allow=all

When I try to call extension 103, I get on the local PBX console (the one
connected directly to the internet):

    -- Executing Dial("SIP/351-d7ca", "IAX/pspbx at n.n.n.n/103") in new stack
    -- Calling using options
'exten=103;callerid="MyOffice"<351>;language=en;username=pspbx;formats=2;cap
ability=65283;version=1;adsicpe=2'
    -- Called pspbx at n.n.n.n/103
  == No one is available to answer at this time
    -- Hungup 'IAX[n.n.n.n:5036]/46'
    -- Executing Hangup("SIP/351-d7ca", "") in new stack
  == Spawn extension (fullaccess, 103, 2) exited non-zero on 'SIP/351-d7ca'

and a busy tone.
Nothing one the console of the other PBX.

The PBX behind the NAT is in DMZ (so completely exposed to the internet).
Making a VPN connection between the two PBX (using PPTP) it works in both
direction, so this is only a NAT related issue.

I have missed something in configuring the system?


Thanks and best regards,
Dan







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