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<DIV><SPAN class=546102920-26112006><FONT face=Arial size=2>Ricardo
-</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial size=2>this is an Asterisk
issue.</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial size=2>Configure your
SIP.conf in * creating a sip peer with insecure=very fotr the SER peer, like
this:</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2></FONT> </DIV></SPAN>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2>[ser-peer]</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial size=2>type =
friend
; should you want to receive and make calls to SER</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2>context=from-ser
; the context in dialplan with extensions allowed to be accessed by SER. Here
you must have PSTN extensions capability.</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2>host=200.XXX.XXX.XXX ; the IP address
for SER</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2>fromdomain=domain.com ;the Domain part of uri to be verified by asterisk
on the INVITE received by SER. </FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial size=2>qualify=
yes
; just to check the latency between SER and Asterisk (like this, if over 2000ms
Ast will report as unavailable peer).</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2>disallow=all</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2>allow=alaw</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2>allow=g729</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial size=2>insecure=
very
; this line garantees that any username part of Request URI sent by SER in
INVITE to Asterisk will be accepted by Ast and routed to the
dialplan.</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial size=2>So, if SER send an
INVITE to <A
href="mailto:5531332818847@domain.com.br">5531332818847@domain.com.br</A> ,
Asterisk will look for a section of type =user in SIP conf to match the
user part first, it won't find, then it will look for a type=peer. It will find
and try to match the IP address as in host= ...line and will accept any username
part as per insecure=very line. Then, if context=from-ser in the Ast dialplan
allows this dialing string (553132818847), it will proceed from
there.</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial size=2>Hope it
helps.</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2>At.</FONT></SPAN></DIV>
<DIV><SPAN class=546102920-26112006><FONT face=Arial
size=2>Walter</FONT></SPAN></DIV></BODY></HTML>
<BR>
<P><FONT SIZE=2>--<BR>
No virus found in this outgoing message.<BR>
Checked by AVG Free Edition.<BR>
Version: 7.5.430 / Virus Database: 268.14.16/552 - Release Date: 26/11/2006 11:30<BR>
</FONT> </P>