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<DIV><SPAN class=479342115-14022006><FONT face=Arial color=#0000ff size=2>If
Asterisk is in the public network, it will work. The problem is when Asterisk is
behind NAT and one of the client is also behind the same
NAT.</FONT></SPAN></DIV>
<BLOCKQUOTE>
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]<B>On Behalf Of
</B>turby<BR><B>Sent:</B> Tuesday, February 14, 2006 6:35 AM<BR><B>To:</B>
'Asterisk Users Mailing List - Non-Commercial Discussion'<BR><B>Subject:</B>
RE: [Asterisk-Users] Dial command to connect two channels and bypassasterisk
server<BR><BR></FONT></DIV>
<DIV dir=ltr align=left><SPAN class=368133311-14022006><FONT face=Arial
color=#0000ff size=2>this is not usefull for public enviroment. clients behind
nat does not work...</FONT></SPAN></DIV>
<DIV><SPAN class=368133311-14022006></SPAN><FONT face=Arial><FONT
color=#0000ff><FONT size=2>turby</FONT></FONT></FONT></DIV>
<DIV><FONT face=Arial><FONT color=#0000ff><FONT size=2><SPAN
class=368133311-14022006></SPAN></FONT></FONT></FONT><BR> </DIV>
<DIV class=OutlookMessageHeader lang=cs dir=ltr align=left>
<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>Nitin
Gupta<BR><B>Sent:</B> Tuesday, February 14, 2006 10:51 AM<BR><B>To:</B>
Asterisk Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[Asterisk-Users] Dial command to connect two channels and bypass asterisk
server<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV>thanks for the information Peter, its really helpful. Also I have
one more question - do you have any idea how many such simultaneous calls can
an asterisk server handle (say running od 2.6Ghz, 1GB Ram, fedora machine)?
</DIV>
<DIV> </DIV>
<DIV>Thanks,</DIV>
<DIV>Nitin<BR><BR> </DIV>
<DIV><SPAN class=gmail_quote>On 2/13/06, <B class=gmail_sendername>Peter
Fern</B> <<A
href="mailto:pete@keypoint.com.au">pete@keypoint.com.au</A>> wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">You
can enable this on a per-peer basis with:<BR><BR>sip
peers:<BR>canreinvite=yes<BR><BR>iax peers:<BR>notransfer=no <BR><BR>Check
the iax.conf.sample and sip.conf.sample files for usage.<BR><BR>Nitin Gupta
wrote:<BR><BR>> Hi I was wondering if its possible to make Dial command
bridge two<BR>> channels and after bridging bypass asterisk, so that the
voice doesn't <BR>> need to pass through my asterisk
server.<BR>><BR>> For e.g., I have a user dialed in and he verifies
himself and then<BR>> dials an international extension, after the call
connects I don't want<BR>> the call to pass through asterisk server
anymore. Is there any command <BR>> already there for any particular
channel type?<BR>><BR>> Thanks,<BR>>
Nitin<BR>><BR>>------------------------------------------------------------------------<BR>><BR>>_______________________________________________
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