<div dir="ltr">Howdy,<div><br></div><div>Your sip.conf file looks fine for some testing, though I would recommend _not_ using an extension number to name a sip endpoint. Instead, name the sip endpoint something more descriptive of the device. [Linphone-01] [Linphone-02] for example. Then you'll want to configure extensions.conf to Dial() the sip endpoint whenever the extension is dialed.</div>
<div><br></div><div><br></div><div class="gmail_extra"><div><div dir="ltr"><div><font face="courier new, monospace" color="#444444">Justin Hester</font></div><div><font face="courier new, monospace" color="#444444">Digium, Inc. · Technical Trainer</font></div>
<div><font face="courier new, monospace" color="#444444">445 Jan Davis Drive NW · Huntsville, AL 35806 · USA</font></div><div><font face="courier new, monospace" color="#444444">ph: +1 256 428 6238</font></div><div><font face="courier new, monospace" color="#444444">Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> · <a href="http://asterisk.org" target="_blank">http://asterisk.org</a></font></div>
</div></div>
<br><br><div class="gmail_quote">On Mon, Feb 3, 2014 at 5:45 AM, Raghav Goud <span dir="ltr"><<a href="mailto:raghavgoud.g@gmail.com" target="_blank">raghavgoud.g@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">Hi all,<div><br></div><div> I want to two sip clients connect through Asterisk in local network for testing. My sip.conf file looks like this </div><div><br></div><div> [general]</div><div>
context=internal</div><div>allowguest=no</div><div>allowoverlap=no</div><div>bindport=5060</div><div>bindaddr=0.0.0.0</div><div>srvlookup=no</div><div>disallow=all</div><div>allow=ulaw</div><div>alwaysauthreject=yes</div>
<div>canreinvite=no</div><div>nat=yes</div><div>session-timers=refuse</div><div>localnet=<a href="http://192.168.1.0/255.255.255.0" target="_blank">192.168.1.0/255.255.255.0</a></div><div><br></div><div>[7001]</div><div>
type=friend</div>
<div>host=dynamic</div><div>secret=123abcd</div><div>context=internal</div><div><br></div><div>[7002]</div><div>type=friend</div><div>host=dynamic</div><div>secret=456abcd</div><div>context=internal</div><div><br></div><div>
<br></div><div>Am using linphone as sip client and create account on linphone with user name 7001 and 7002 </div><div>7001 is running on <a href="http://192.168.2.15:5060" target="_blank">192.168.2.15:5060</a></div><div>
7002 is running on <a href="http://192.168.2.45:5060" target="_blank">192.168.2.45:5060</a> </div><div><br></div><div>when i try to call from 7002 to 7001 i specified <a href="mailto:sip%3A7001@192.168.2.15" target="_blank">sip:7001@192.168.2.15</a> it working fine as i know ip adress i specified it as url. if i dnt know the ipadress how can i call to 7001? i try to call <a href="mailto:sip%3A7001@192.168.2.20" target="_blank">sip:7001@192.168.2.20</a> it through call rejected because extension not found in context 'internal, error.</div>
<div> </div><div> How can call to sip id with out knowning ipadress where it is runnning? Any modification required for sip.conf file?</div><div><br></div><div>Thanks,</div><div>Raghav</div>
<div> </div><div><br></div><div> </div></div>
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