[Users] Re: [Devel] openser and asterisk

Matt L. Zhu coder0000 at hotmail.com
Thu Sep 29 20:22:04 CEST 2005

in my dialplan, i have this

[proxy]    # same as the context in sip.conf
exten => 4005.,1,Dial(SIP/${EXTEN}@

i am new to asterisk, how can i make it so the exten will route the call to 
the other sipphone connected to the ser proxy.

i really want to achieve sipphone->ser->asterisk->ser->sipphone when a phone 
calls another. just getting confused how exten will reroute to ser again.

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From:  <i>Mark Aiken &lt;aiken.mark at gmail.com&gt;</i><br>Reply-To:  <i>Mark 
Aiken &lt;aiken.mark at gmail.com&gt;</i><br>To:  <i>Iqbal 
&lt;iqbal at gigo.co.uk&gt;</i><br>CC:  <i>Bogdan-Andrei Iancu 
&lt;bogdan at voice-system.ro&gt;, &quot;Matt L. Zhu&quot; 
&lt;coder0000 at hotmail.com&gt;, users at openser.org</i><br>Subject:  <i>Re: 
[Users] Re: [Devel] openser and asterisk</i><br>Date:  <i>Thu, 29 Sep 2005 
12:04:21 -0500</i><br>
<br>You may want to set type=peer in the [ser] section. Also , I assume you
have a Dial statement in your 'proxy' context in the dialplan. You need
that to connect the 2 users. We have no problems using Asterisk as a
sip server with ser or openser as the registrar and proxy. I think
there are many using this kind of setup so it does work.<br>
Mark<br><br><div><span class="gmail_quote">On 9/29/05, <b 
class="gmail_sendername">Iqbal</b> &lt;<a 
href="mailto:iqbal at gigo.co.uk">iqbal at gigo.co.uk</a>&gt; 
wrote:</span><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 
whats is sip debug on asterisk showing<br><br>Bogdan-Andrei Iancu 
wrote:<br><br>&gt; Hi Matt,<br>&gt;<br>&gt; I redirected this email on the 
users mailing list - it's more<br>&gt; appropriate.<br>&gt;<br>&gt; the idea 
seams ok, with couple of comments:
<br>&gt;    1) be sure that fwd to localhost is ok (instead of a routable 
IP)<br>&gt;    2) doing Record-Route may be a good think.<br>&gt;<br>&gt; to 
debug tour problem, add some log(&quot;...&quot;) statements into your 
<br>&gt; to be able to trace the processing. Also a network trace (including 
on<br>&gt; lo device) will be helpful to see what happens - if the messages 
are<br>&gt; received, if they are sent and where. Also watch the log for 
<br>&gt; errors.<br>&gt;<br>&gt; regards,<br>&gt; 
bogdan<br>&gt;<br>&gt;<br>&gt;<br>&gt; Matt L. Zhu 
wrote:<br>&gt;<br>&gt;&gt; has anyone successfully setup openser as the 
frontend proxy for<br>&gt;&gt; asterisk? here is my setup
<br>&gt;&gt;<br>&gt;&gt; /etc/asterisk/sip.conf<br>&gt;&gt; 
[general]<br>&gt;&gt; context=default<br>&gt;&gt; port=5065<br>&gt;&gt; 
bindaddr=<a href=""></a><br>&gt;&gt; 
srvlookup=yes<br>&gt;&gt;<br>&gt;&gt; [ser]
<br>&gt;&gt; type=user<br>&gt;&gt; context=proxy<br>&gt;&gt; host=<a 
href=""></a><br>&gt;&gt;<br>&gt;&gt; then i 
edited openser.cfg to do something like this<br>&gt;&gt;<br>&gt;&gt;         
{<br>&gt;&gt;                    forward(
localhost, 5065 );<br>&gt;&gt;                    break;<br>&gt;&gt;         
    };<br>&gt;&gt;<br>&gt;&gt; i connected two sipphones (wengo) in this 
case to openser, but calls<br>&gt;&gt; are not going through at all, 
connecting directly to asterisk works.
<br>&gt;&gt; have anyone worked in this situation?<br>&gt;&gt;<br>&gt;&gt; 
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