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thanks Samy<br>
<br>
i will do the test in the following coming days.<br>
<br>
Michael<br>
<br>
<br>
<div class="moz-cite-prefix">On 11/04/2014 12:15 AM, SamyGo wrote:<br>
</div>
<blockquote
cite="mid:CAJUJwtjq=49wqynggu3omixKigCn_emqxk53G2GERdB8reqs=Q@mail.gmail.com"
type="cite">
<div dir="ltr">Hi Michael,
<div><br>
<div>You can follow what Bogdan mentioned and your call will
get routed to the other OpenSIPS as long as the FQDN is
resolvable. If your users don't want to add the other domain
while dialling then you can add the static routing code
logic in the opensips.cfg file and restart opensips. </div>
</div>
<div><br>
</div>
<div>Thanks,</div>
<div>Sammy</div>
<div><br>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Mon, Nov 3, 2014 at 5:06 AM, Michael
Leung <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:gbcbooksmj@gmail.com" target="_blank">gbcbooksmj@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">thanks for you all reply
<div><br>
</div>
<div>now i have a further knowing of SIP</div>
<div><br>
</div>
<div>to Samy </div>
<div><br>
</div>
<div>which configure file should i add these syntax to ? </div>
<div><br>
</div>
<div>
<div>root@CDlinux:/home/yliang1# ls
/usr/local/opensips_proxy/etc/opensips/opensips_residential_2014-10-26_15\:6\:17.cfg</div>
<div><br>
</div>
<div><b>/usr/local/opensips_proxy/etc/opensips/opensips_residential_2014-10-26_15:6:17.cfg</b></div>
<div><b><br>
</b></div>
<div>root@CDlinux:/home/yliang1# </div>
</div>
<div><br>
</div>
<div><br>
</div>
<div>i followed the tutorial to install opensips to
/usr/local/opensips_proxy</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</div>
<div class="HOEnZb">
<div class="h5">
<div class="gmail_extra"><br>
<div class="gmail_quote">On Mon, Nov 3, 2014 at 2:26
PM, Bogdan-Andrei Iancu <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:bogdan@opensips.org"
target="_blank">bogdan@opensips.org</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div><tt>Hi Michael,<br>
<br>
In SIP, each SIP server does server a
certain set of SIP domains (defined as FQDNs
or IPs). Let's assume phone A registers with
proxy A using domainA and </tt><tt><tt>phone
B registers with proxy B using domainB. <br>
To have a call from phone A to phone B, A
must dial B@domainB - so when the call
will land on proxy A, proxy A will see
domainB is not locally served and it will
forward to proxy B (responsible for
domainB). This is call inter-domain DNS
based routing and it is covered by the
OpenSIPS default script.<br>
<br>
</tt>Regards,<br>
</tt>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<div>
<div> On 31.10.2014 13:13, Michael Leung
wrote:<br>
</div>
</div>
</div>
<blockquote type="cite">
<div>
<div>
<div dir="ltr">Hi all
<div><br>
</div>
<div>i know this is a stupid question</div>
<div><br>
</div>
<div>but i dont use sip to make a phone
call very often , </div>
<div><br>
</div>
<div>i have setup up two opensips server
in my intranet environment</div>
<div><br>
</div>
<div>i use two phones to register on
each server </div>
<div><br>
</div>
<div>how to make a phone call from one
to another one</div>
<div><br>
</div>
<div>do i have to add the the
destination domain name behind the
alias number when i dial out ?</div>
<div><br>
</div>
<div>or why can i dial the alias number
without domain name , then the
opensips server will routing it to a
the opensips server automatically </div>
<div><br>
</div>
<div><br>
</div>
<div>thanks </div>
<div><br>
</div>
<div>Michael</div>
</div>
<br>
<fieldset></fieldset>
<br>
</div>
</div>
<span>
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</span></blockquote>
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</blockquote>
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</div>
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