[Asterisk-Users] HT-486 Voice Nat Problem

[Alvaro Parres]

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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