<p dir="ltr">Thanks for the help. </p>
<p dir="ltr">Is it possible to have a direct sip trunk from kamailio to an NGN without involving asterisks? I will be using NGN to route outside calls for landlines and mobile. </p>
<div class="gmail_quote">On Mar 24, 2014 9:37 PM, "Rainer Piper" <<a href="mailto:rainer.piper@soho-piper.de">rainer.piper@soho-piper.de</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div>Hi Rizwan,<br>
<br>
that is the right approach .<br>
<br>
For adding an Asterisk as SBC you can use the section route[PSTN]
at kamailio.cfg<br>
<br>
#!ifdef WITH_PSTN<br>
# PSTN GW Routing<br>
#<br>
# - pstn.gw_ip: valid IP or hostname as string value, example:<br>
# pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address"<br>
#<br>
# - by default is empty to avoid misrouting<br>
pstn.gw_ip = "IP_OF_YOUR_ASTERISK"<br>
pstn.gw_port = "5060"<br>
#!endif<br>
<br>
Make kamailio IP as trusted IP in your asterisk sip.conf like<br>
<br>
[kamailio]<br>
type=friend<br>
context=outgoing-kamailio<br>
host=[IP_OF_YOUR_KAMAILIO]<br>
port=5060<br>
qualify=no<br>
;trustrpid=yes<br>
;sendrpid=yes<br>
deny=<a href="http://0.0.0.0/0.0.0.0" target="_blank">0.0.0.0/0.0.0.0</a><br>
permit=[IP_OF_YOUR_KAMAILIO]<br>
<br>
add outgoing trunk Data to your asterisk sip.conf and
extensions.conf section [outgoing-kamailio]<br>
<br>
and that is it.<br>
<br>
<br>
Regards<br>
Rainer<br>
<br>
<br>
<br>
<br>
Am 24.03.2014 16:23, schrieb Rizwan Khan:<br>
</div>
<blockquote type="cite">
<p dir="ltr">Is my question not well phrased? Or is too general?
Can anyone help with a document or an older thread which could
help me? </p>
<p dir="ltr">Thanks </p>
<div class="gmail_quote">On Mar 24, 2014 1:57 PM, "Rizwan Khan"
<<a href="mailto:rizkhan@gmail.com" target="_blank">rizkhan@gmail.com</a>>
wrote:<br type="attribution">
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div>I want the following setup:</div>
<div><br>
</div>
<div>1 Kamailio server to handle internal calls (A/V), IM
and Presence.</div>
<div>1 Asterisk or any other way to communicate with an NGN
where I will create the SIP Trunk to route calls outside
of the network.</div>
<div><br>
</div>
<div>Is this the right approach or there is a way to
directly communicate with the NGN to make a SIP trunk by
using some external modules. </div>
<div><br>
</div>
<div>Any guidelines will be highly appreciated.</div>
<div><br>
</div>
<br clear="all">
<div>Rizwan Khan<br>
<br>
<br>
</div>
</div>
</blockquote>
</div>
<br>
<fieldset></fieldset>
<br>
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</blockquote>
<br>
<br>
<div>-- <br>
<b>Rainer Piper</b>
<br>
NOC - <a href="tel:%2B49%20%280%29228%2097167161" value="+4922897167161" target="_blank">+49 (0)228 97167161</a> - <a href="http://sip.soho-piper.de" target="_blank">sip.soho-piper.de</a>
<br>
NOC - <a href="tel:%2B49%20%280%292247%209064188" value="+4922479064188" target="_blank">+49 (0)2247 9064188</a> - <a href="http://sip.tele33.de" target="_blank">sip.tele33.de</a> - <a href="http://sip.tefonix.de" target="_blank">sip.tefonix.de</a> - D293
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