asterisk users - Nov 2005 - HT-486 Voice Nat Problem

Hi list, i have the next situation:
  B
 [HT486] ------ (NAT/ROUTER) ------------ INTERNET ----------------- [*
server]
 |
 |
 |
 A |
 [HT486] -----------------
 Both HT486 register to * server, with no problem, but when they call each
other
the voice only goes from B to A but not from A to B.
 My configs
 sip.conf
 [B]
type=friend
context=ATAs
callerid=<113>
host=dynamic ; we have a static but private IP address
nat=yes
qualify=no ; there is not NAT between phone and
canreinvite=no ; allow RTP voice traffic to bypass Asterisk
dtmfmode=inband ; either RFC2833 or INFO for the BudgeTone
;incominglimit=2 ; permit only 1 outgoing call at a time
disallow=all ; need to disallow=all before we can use
allow=ulaw ; Note: In user sections the order of codecs
allow=alaw
secret=xxxxxx
group=1
  [A]
type=friend
context=ATAs
callerid=<113>
host=dynamic ; we have a static but private IP address
nat=yes
qualify=no ; there is not NAT between phone and
canreinvite=no ; allow RTP voice traffic to bypass Asterisk
dtmfmode=inband ; either RFC2833 or INFO for the BudgeTone
;incominglimit=2 ; permit only 1 outgoing call at a time
disallow=all ; need to disallow=all before we can use
allow=ulaw ; Note: In user sections the order of codecs
allow=alaw
secret=xxxxxx
group=1
 At the HT486 i have the next:
 SIP SERVER=xxx.xxx.xxx.xxx.
NAT Traversal: = yes IP:stun.xten.org <http://stun.xten.org>
 any idea or help ??
 thanks
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Alvaro, I don't have a solution but a question, if A is not behind NAT why 
do you have nat=yes for it?
Soner

----- Original Message ----- 
From: "Alvaro Parres" <aparres@gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users@lists.digium.com>
Sent: Tuesday, November 01, 2005 10:10 PM
Subject: [Asterisk-Users] HT-486 Voice Nat Problem


Hi list, i have the next situation:
  B
 [HT486] ------ (NAT/ROUTER) ------------ INTERNET ----------------- [*
server]
 |
 |
 |
 A |
 [HT486] -----------------
 Both HT486 register to * server, with no problem, but when they call each
other
the voice only goes from B to A but not from A to B.
 My configs
 sip.conf
 [B]
type=friend
context=ATAs
callerid=<113>
host=dynamic ; we have a static but private IP address
nat=yes
qualify=no ; there is not NAT between phone and
canreinvite=no ; allow RTP voice traffic to bypass Asterisk
dtmfmode=inband ; either RFC2833 or INFO for the BudgeTone
;incominglimit=2 ; permit only 1 outgoing call at a time
disallow=all ; need to disallow=all before we can use
allow=ulaw ; Note: In user sections the order of codecs
allow=alaw
secret=xxxxxx
group=1
  [A]
type=friend
context=ATAs
callerid=<113>
host=dynamic ; we have a static but private IP address
nat=yes
qualify=no ; there is not NAT between phone and
canreinvite=no ; allow RTP voice traffic to bypass Asterisk
dtmfmode=inband ; either RFC2833 or INFO for the BudgeTone
;incominglimit=2 ; permit only 1 outgoing call at a time
disallow=all ; need to disallow=all before we can use
allow=ulaw ; Note: In user sections the order of codecs
allow=alaw
secret=xxxxxx
group=1
 At the HT486 i have the next:
 SIP SERVER=xxx.xxx.xxx.xxx.
NAT Traversal: = yes IP:stun.xten.org <http://stun.xten.org>
 any idea or help ??
 thanks



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