[Asterisk-Users] Budgetone and NAT not working

Tue May 24 14:00:52 MST 2005

I have a couple of Budgetones that I am playing with trying to get them
to work with * from a remote network over the Internet (yes NAT joy!).
My * server is in my DMZ and I have 5060 and my RTP range forwarded
(UDP) to my public address (through a Cisco PIX).  Internally, I can
setup my budgetone, it registers and works great.  I then have a Linksys
router connected to another Internet connection.  When I plug the
budgetone into the linksys, login to it and update the SIP Server
setting to the public IP of my * server, it will not register; I get a
403 Forbidden.  I have changed the NAT setting to Yes and am using a
public STUN server.  

My setup is as follows:

Asterisk Server: 192.168.20.10

Linksys Inside:  192.168.111.0/24

Linksys Outside: 216.###.###.60

When I enable SIP Debug in Asterisk, this is what I get:

Sip read: 

REGISTER sip:192.168.21.10 SIP/2.0

Via: SIP/2.0/UDP 216.###.###.60:28249;branch=z9hG4bK3cf4300cb012236e

From: "Budgetone2"
<sip:402 at 192.168.21.10;user=phone>;tag=4fc66b25585eaa82

To: <sip:402 at 192.168.21.10;user=phone>

Contact: *

Call-ID: d9222911de3fdda0 at 192.168.111.101

CSeq: 100 REGISTER

Expires: 0

User-Agent: Grandstream BT100 1.0.6.2

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE

Content-Length: 0

12 headers, 0 lines

Using latest request as basis request

Sending to 216.###.###.60 : 28249 (non-NAT)

Transmitting (NAT):

SIP/2.0 100 Trying

Via: SIP/2.0/UDP
216.###.###.60:28249;branch=z9hG4bK3cf4300cb012236e;received=216.###.###
.60;rport=28249

From: "Budgetone2"
<sip:402 at 192.168.21.10;user=phone>;tag=4fc66b25585eaa82

To: <sip:402 at 192.168.21.10;user=phone>;tag=as7fe61dbd

Call-ID: d9222911de3fdda0 at 192.168.111.101

CSeq: 100 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:402 at 192.168.21.10>

Content-Length: 0

 to 216.###.###.60:28249

Transmitting (NAT):

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP
216.###.###.60:28249;branch=z9hG4bK3cf4300cb012236e;received=216.###.###
.60;rport=28249

From: "Budgetone2"
<sip:402 at 192.168.21.10;user=phone>;tag=4fc66b25585eaa82

To: <sip:402 at 192.168.21.10;user=phone>;tag=as7fe61dbd

Call-ID: d9222911de3fdda0 at 192.168.111.101

CSeq: 100 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:402 at 192.168.21.10>

WWW-Authenticate: Digest realm="asterisk", nonce="4feb882d"

Content-Length: 0

 to 216.###.###.60:28249

Scheduling destruction of call 'd9222911de3fdda0 at 192.168.111.101' in
15000 ms

asterisk1*CLI> 

Sip read: 

REGISTER sip:192.168.21.10 SIP/2.0

Via: SIP/2.0/UDP 216.###.###.60:28249;branch=z9hG4bKe702db9832e47e6b

From: "Budgetone2"
<sip:402 at 192.168.21.10;user=phone>;tag=4fc66b25585eaa82

To: <sip:402 at 192.168.21.10;user=phone>

Contact: *

Authorization: DIGEST username="402", realm="asterisk", algorithm=MD5,
uri="sip:192.168.21.10", nonce="4feb882d",
response="83dd6741f472e9690ca207d385cb27f0"

Call-ID: d9222911de3fdda0 at 192.168.111.101

CSeq: 101 REGISTER

Expires: 0

User-Agent: Grandstream BT100 1.0.6.2

Max-Forwards: 70

Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE

Content-Length: 0

13 headers, 0 lines

Using latest request as basis request

Sending to 216.###.###.60 : 28249 (NAT)

Transmitting (NAT):

SIP/2.0 100 Trying

Via: SIP/2.0/UDP
216.###.###.60:28249;branch=z9hG4bKe702db9832e47e6b;received=216.###.###
.60;rport=28249

From: "Budgetone2"
<sip:402 at 192.168.21.10;user=phone>;tag=4fc66b25585eaa82

To: <sip:402 at 192.168.21.10;user=phone>;tag=as7fe61dbd

Call-ID: d9222911de3fdda0 at 192.168.111.101

CSeq: 101 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:402 at 192.168.21.10>

Content-Length: 0

 to 216.###.###.60:28249

Transmitting (NAT):

SIP/2.0 403 Forbidden

Via: SIP/2.0/UDP
216.###.###.60:28249;branch=z9hG4bKe702db9832e47e6b;received=216.###.###
.60;rport=28249

From: "Budgetone2"
<sip:402 at 192.168.21.10;user=phone>;tag=4fc66b25585eaa82

To: <sip:402 at 192.168.21.10;user=phone>;tag=as7fe61dbd

Call-ID: d9222911de3fdda0 at 192.168.111.101

CSeq: 101 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:402 at 192.168.21.10>

Content-Length: 0

 to 216.###.###.60:28249

Scheduling destruction of call 'd9222911de3fdda0 at 192.168.111.101' in
15000 ms

So the Grandstreams will not work...not matter what I try.  However, I
have XLite installed on my home computer and when I attempt to connect
over the Internet with that, it works!  The only difference in the
config that I can see is that in XLite you can set your Domain/Realm.
In the budgetone, I can not.  I'm running version 1.0.6.2 firmware in
the budgetone.

Please let me know if you have any suggestions.  Thanks in advance.

Dan

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