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" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
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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