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<DIV dir=ltr align=left><SPAN class=578445014-23112005><FONT face=Arial
color=#0000ff size=2>I'm pasting something from another user on this list from
14/11/05</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=578445014-23112005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=578445014-23112005>
<P><FONT size=2>I would recommend that you do a little research on google, voip-
info.org, and the list archives.</FONT></P>
<P><FONT size=2>To connect to an Asterisk box that sits behind NAT, you need to
forward ports 5060 and 10000-20000 too the asterisk box, and you need to
configure the externip, localnet, and nat variables in sip.conf. </FONT></P>
<P><FONT size=2>audio problems are almost always due to the RTP stream
(ports<SPAN class=578445014-23112005> </SPAN></FONT><FONT size=2>10000-20000)
not being forwarded properly, either due to the port forwarding setup or the
sip.conf settings.</FONT></P>
<P><FONT size=2>Tom</FONT></P>
<P><FONT
size=2>----------------------------------------------------------</FONT></P>
<P><FONT size=2>Tom Rymes</FONT></P>
<P><FONT size=2>Cascade Link Systems</FONT></P>
<P><A href="outbind://12/www.cascadelinksystems.com"><U><FONT color=#0000ff
size=2>www.cascadelinksystems.com</U></FONT></A></P><FONT size=2>
<P>(603) 375-1414</P></FONT></SPAN></DIV><BR>
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<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Bharath
Khambadkone<BR><B>Sent:</B> Wednesday, November 23, 2005 9:29 AM<BR><B>To:</B>
Asterisk Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[Asterisk-Users] SIP Extension behind NAT,Asterisk on a public
domain<BR></FONT><BR></DIV>
<DIV></DIV>By default AMP had NAT=yes in sip.conf, I read in some posts to
change it to one, i was just trying my luck if that works. I have tried NAT=yes,
The Phone gets registered, I can also make & recieve calls but as soon as
the call is picked I dont hear anything at both ends. Does this have anything to
do with codecs?<BR><BR>Thanks<BR><BR>
<DIV><SPAN class=gmail_quote>On 11/22/05, <B class=gmail_sendername>C F</B>
<<A href="mailto:shmaltz@gmail.com">shmaltz@gmail.com</A>> wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">On
11/22/05, Bharath Khambadkone <<A
href="mailto:bkalthod@gmail.com">bkalthod@gmail.com</A>> wrote:<BR>>
Hello All,<BR>> I'm fairly new to asterisk. I have read about
the problems about NAT, But<BR>> can't seem to find a solution.
<BR>> My Asterisk is on a public domain, there is no NAT or
firewall in front of<BR><BR><BR>If no nat then why do you have nat=1 in
sip.conf?<BR><BR><BR>> the asteris box. I have sucessfully connected iax2
softphones & was able to <BR>> recieve & make calls. In the same
locations where I have the iax2 extensions<BR>> working I have set up a a
SIP softphone & a SIP ATA (Sipura2002). Both teh<BR>> sip phones are
able to register. I can also make & recieve calls but cannot <BR>> hear
anything after the call is answered at both ends. I'm not sure what is<BR>>
causing this problem. By the way I'm using SME server 7(centos
4.2) with<BR>> A@H installed.<BR>><BR>> my
Sip.conf :<BR>> [2008]
;(Sipura2002)<BR>> username=2008<BR>> type=friend<BR>> secret=2008<BR>> record_out=Adhoc<BR>> record_in=Adhoc<BR>> qualify=no<BR>> port=5060<BR>> nat=1<BR>> mailbox=2008@device
<BR>> host=dynamic<BR>> dtmfmode=rfc2833<BR>> context=from-internal<BR>> canreinvite=no<BR>> callerid=device
<2008><BR>><BR>><BR>> [2009] ;X-Lite Soft
Phone<BR>> username=2009<BR>> type=friend
<BR>> secret=2009<BR>> record_out=Adhoc<BR>> record_in=Adhoc<BR>> qualify=no<BR>> port=5060<BR>> nat=1<BR>> mailbox=2009@device<BR>> host=dynamic<BR>> dtmfmode=rfc2833<BR>> context=from-internal
<BR>> canreinvite=no<BR>> callerid=device
<2009><BR>><BR>> Thanks in
advance..<BR>><BR>><BR>><BR>><BR>><BR>>
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