<div dir="ltr"><div class="gmail_default" style="color:#660000">It would appear that you have the Asterisk server on a public IP address, your two endpoints are behind a NAT, and you have rewrite_contact enabled in pjsip.conf.</div><div class="gmail_default" style="color:#660000"><br></div><div class="gmail_default" style="color:#660000">In which case, what you are seeing is correct. In order to be able to send a call to an extension where it is behind NAT, Asterisk must update the contact to have the current IP and port that the phone registered via (i.e. the WAN IP of the NAT, and the WAN port that it is retaining state for).</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan <span dir="ltr"><<a href="mailto:sonny.rajagopalan@gmail.com" target="_blank">sonny.rajagopalan@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"><div>I am following the instructions in <a href="https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality</a> and I am trying to make a call from extension Alice (6001) to extension for Bob (6002). When I make the call, I can hear the ringing on Alice's phone (caller), but Bob's phone (callee) doesn't ring, or show a call coming in from Alice. My setup and environment is as follows: Alice, Bob and Asterisk all in the same <a href="http://192.168.1.0/24" target="_blank">192.168.1.0/24</a> network, and they are able to register to the Asterisk server running 13.1.0/PJSIP. The rest of the configuration is the same as the aforementioned wiki page, but is shown here for clarity:</div><div><br></div><blockquote style="margin:0 0 0 40px;border:none;padding:0px"><font color="#ff0000" size="1" face="monospace, monospace">root@asterisk13FFP:/var/log/asterisk# more /etc/asterisk/extensions.conf</font><font color="#ff0000" size="1" face="monospace, monospace"><br></font><font color="#ff0000" size="1" face="monospace, monospace">[from-internal]</font><font color="#ff0000" size="1" face="monospace, monospace"><br></font><font color="#ff0000" size="1" face="monospace, monospace">exten=>6001,1,Dial(PJSIP/demo-alice)</font><font color="#ff0000" size="1" face="monospace, monospace"><br></font><font color="#ff0000" size="1" face="monospace, monospace">exten=>6002,1,Dial(PJSIP/demo-bob)</font><font color="#ff0000" size="1" face="monospace, monospace"><br></font><font color="#ff0000" size="1" face="monospace, monospace">exten=>6003,1,Answer()</font><font color="#ff0000" size="1" face="monospace, monospace"><br></font><font color="#ff0000" size="1" face="monospace, monospace">same =>6003,n,Playback(hello-world)</font><font color="#ff0000" size="1" face="monospace, monospace"><br></font><font color="#ff0000" size="1" face="monospace, monospace">same =>6003,n,Hangup() </font></blockquote><div><br></div><div>What I do observe is that I when I request the output of <font face="monospace, monospace" color="#ff0000">pjsip show endpoints</font>, I get Contact information for the two SIP peers that have registered different from their actual IP addresses. I suspect this has something to do with their calls being routed elsewhere. If my assumption is correct--how do I fix this? Alice should be at 192.168.1.50 and Bob should be at 192.168.1.149, instead, they (both) show IP address 146.115.163.234. Any help is deeply appreciated. Thanks.</div><div><br></div><blockquote style="margin:0px 0px 0px 40px;border:none;padding:0px"><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000">asterisk13FFP*CLI> pjsip show endpoints</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"><br></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> I/OAuth: <AuthId/UserName...........................................................></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Aor: <Aor............................................> <MaxContact></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Contact: <Aor/ContactUri...............................> <Status....> <RTT(ms)..></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Identify: <Identify/Endpoint.........................................................></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Match: <ip/cidr.........................></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Channel: <ChannelId......................................> <State.....> <Time(sec)></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> =========================================================================================</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"><br></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Endpoint: demo-alice Unavailable 0 of inf</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> InAuth: demo-alice/demo-alice</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Aor: demo-alice 1</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Contact: demo-alice/sip:demo-alice@<b>146.115.163.234</b>:38519 Unknown nan</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"><br></font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Endpoint: demo-bob Not in use 0 of inf</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> InAuth: demo-bob/demo-bob</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Aor: demo-bob 1</font></div><div><font face="monospace, monospace" size="1" style="background-color:rgb(255,255,255)" color="#ff0000"> Contact: demo-bob/sip:demo-bob@<b>146.115.163.234</b>:38321;tra Unknown nan</font></div></blockquote></div>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature"><div dir="ltr"><img alt="Digium logo" src="https://my.digium.com/images/graphics/digium_RGB_signature.gif" width="288" height="50" style="color:rgb(0,0,0);font-family:Arial,Helvetica,sans-serif;font-size:12px"><div>Scott Griepentrog<br>Digium, Inc · Software Developer<br>445 Jan Davis Drive NW · Huntsville, AL 35806 · US<br>direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090<br>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><br></div></div></div>
</div>