<div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi Jerry &amp; List,</blockquote><div><br>
<br>
I have the following registrations&nbsp; in sip_additional.conf<br>
<br>
register=02820XXXX:&lt;correct password&gt;@202.177.XXX.XXX/02820XXXX<br>
<br>
[02820XXXX]<br>
type=user<br>
secret=&lt;correct password&gt;<br>
host=202.177.XXX.XXX<br>
context=from-pstn<br>
<br>
sip_additional.conf is (or should be) included from sip.conf<br>
<br>
Any other suggestions? Unfortuantely I wasn't backing up my conf files before this happened.<br>
<br>
Scott<br>
<br>
 </div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">------------------------------<br><br>Message: 15<br>Date: Thu, 24 Nov 2005 00:17:06 -0600
<br>From: Jerry Jones &lt;<a href="mailto:jjones@danrj.com">jjones@danrj.com</a>&gt;<br>Subject: Re: [Asterisk-Users] Loss of Registration for SIP Trunks<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;
<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>&gt;<br>Message-ID: &lt;<a href="mailto:4FA9B4B0-3711-4B89-B590-DE7B5FCA5D49@danrj.com">4FA9B4B0-3711-4B89-B590-DE7B5FCA5D49@danrj.com</a>
&gt;<br>Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed<br><br><br>On Nov 24, 2005, at 12:06 AM, Scott Clements wrote:<br><br>&gt; HI List,<br>&gt;<br>&gt; You'll have to pardon the newbieness of this question, I was
<br>&gt; editing the sip.conf file on my asterisk server yesterday, and now<br>&gt; none of my asterisk trunks will connect. From my knowledge sip.conf<br>&gt; does not effect registration, but there have been no other changes
<br>&gt; at all. Below is my sip.conf, and some other CLI info. If anone has<br>&gt; some thoughts please let me know.<br>&gt;<br>&gt;<br>&gt; [general]<br>&gt;<br>&gt; port = 5060&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ; Port to bind to (SIP is 5060)
<br>&gt; bindaddr = <a href="http://0.0.0.0">0.0.0.0</a>&nbsp;&nbsp;&nbsp;&nbsp;; Address to bind to (all addresses on machine)<br>&gt; disallow=all<br>&gt; allow=g729<br>&gt; allow=ulaw<br>&gt; allow=alaw<br>&gt; context=from-pstn<br>&gt; ;context = from-sip-external ; Send unknown SIP callers to this
<br>&gt; context<br>&gt; callerid = Unknown<br>&gt; ;dtmfmode=rfc2833<br>&gt; ;relaxdtmf=yes<br>&gt;<br>&gt; #include sip_nat.conf<br>&gt; #include sip_custom.conf<br>&gt; #include sip_additional.conf<br>&gt;<br>&gt;<br>&gt;
<br>&gt;<br>&gt; cee*CLI&gt; sip show registry<br>&gt;
Host&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Username&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Refresh State<br>This shows other servers to which asterisk has registered. I see no<br>register statements in your sip.conf above.<br>&gt;<br>&gt;<br>&gt; cee*CLI&gt; sip show peers<br>&gt;
Name/username&nbsp;&nbsp;&nbsp;&nbsp;Host&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Dyn
Nat ACL Mask<br>&gt; Port&nbsp;&nbsp;&nbsp;&nbsp; Status<br>&gt;
sip-out-test/02&nbsp;&nbsp;<a href="http://202.177.222.24">202.177.222.24</a>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href="http://255.255.255.255">255.255.255.255</a><br>&gt; 5060&nbsp;&nbsp;&nbsp;&nbsp; Unmonitored<br>&gt;
127/127&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;(Unspecified)&nbsp;&nbsp;&nbsp;&nbsp;D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href="http://255.255.255.255">255.255.255.255</a><br>&gt; 0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Unmonitored<br>&gt;
126/126&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;(Unspecified)&nbsp;&nbsp;&nbsp;&nbsp;D&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href="http://255.255.255.255">255.255.255.255</a><br>&gt; 0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Unmonitored<br>These are sip clients registered to your asterisk server. I see no<br>users listed in your 
sip.conf above, though I guess they are in your<br>include files. I also looks like user sip-out-test has a hardcoded IP<br>and is not set to dynamic so cannot really tell if it is registered<br>or not from this info. Users 127 and 126 are not registered. None
<br>have a qualify to verify connectivity.<br><br>perhaps restoring to your previous config and editing more slowly<br>will show where things broke:)<br><br>&gt;<br>&gt;<br>&gt;<br>&gt;<br>&gt; I have tried removing the trunks, confirmed the username and
<br>&gt; passwords for the trunks are ok. I am totally stumped as to what<br>&gt; would cause it.<br>&gt;<br>&gt; If anyone can help it'd be great :)<br>&gt;<br>&gt; SCott<br>&gt; _______________________________________________
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</blockquote></div><br>