in my dialplan, i have this
[proxy] # same as the context in sip.conf exten => 4005.,1,Dial(SIP/${EXTEN}@192.168.0.10)
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.
<BLOCKQUOTE style='PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #A0C6E5 2px solid; MARGIN-RIGHT: 0px'><font style='FONT-SIZE:11px;FONT-FAMILY:tahoma,sans-serif'><hr color=#A0C6E5 size=1> From: <i>Mark Aiken <aiken.mark@gmail.com></i><br>Reply-To: <i>Mark Aiken <aiken.mark@gmail.com></i><br>To: <i>Iqbal <iqbal@gigo.co.uk></i><br>CC: <i>Bogdan-Andrei Iancu <bogdan@voice-system.ro>, "Matt L. Zhu" <coder0000@hotmail.com>, users@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> <br> Mark<br><br><div><span class="gmail_quote">On 9/29/05, <b class="gmail_sendername">Iqbal</b> <<a href="mailto:iqbal@gigo.co.uk">iqbal@gigo.co.uk</a>> wrote:</span><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;padding-left:1ex"> whats is sip debug on asterisk showing<br><br>Bogdan-Andrei Iancu wrote:<br><br>> Hi Matt,<br>><br>> I redirected this email on the users mailing list - it's more<br>> appropriate.<br>><br>> the idea seams ok, with couple of comments: <br>> 1) be sure that fwd to localhost is ok (instead of a routable IP)<br>> 2) doing Record-Route may be a good think.<br>><br>> to debug tour problem, add some log("...") statements into your script <br>> to be able to trace the processing. Also a network trace (including on<br>> lo device) will be helpful to see what happens - if the messages are<br>> received, if they are sent and where. Also watch the log for potential <br>> errors.<br>><br>> regards,<br>> bogdan<br>><br>><br>><br>> Matt L. Zhu wrote:<br>><br>>> has anyone successfully setup openser as the frontend proxy for<br>>> asterisk? here is my setup <br>>><br>>> /etc/asterisk/sip.conf<br>>> [general]<br>>> context=default<br>>> port=5065<br>>> bindaddr=<a href="http://0.0.0.0">0.0.0.0</a><br>>> srvlookup=yes<br>>><br>>> [ser] <br>>> type=user<br>>> context=proxy<br>>> host=<a href="http://192.168.0.10">192.168.0.10</a><br>>><br>>> then i edited openser.cfg to do something like this<br>>><br>>> if <br>>> (uri=~"sip:[a-zA-Z.]*@(xxx.xxx.com)|(192.168.0.10)") {<br>>> forward( localhost, 5065 );<br>>> break;<br>>> };<br>>><br>>> i connected two sipphones (wengo) in this case to openser, but calls<br>>> are not going through at all, connecting directly to asterisk works. <br>>> have anyone worked in this situation?<br>>><br>>> thanks<br>>><br>>><br>>><br>>> _______________________________________________<br>>> Devel mailing list<br>>> <a href="mailto:Devel@openser.org">Devel@openser.org</a><br>>> <a href="http://openser.org/cgi-bin/mailman/listinfo/devel">http://openser.org/cgi-bin/mailman/listinfo/devel</a><br>>><br>><br>><br>> _______________________________________________ <br>> Users mailing list<br>> <a href="mailto:Users@openser.org">Users@openser.org</a><br>> <a href="http://openser.org/cgi-bin/mailman/listinfo/users">http://openser.org/cgi-bin/mailman/listinfo/users</a><br>> <br>> .<br>><br><br>_______________________________________________<br>Users mailing list<br><a href="mailto:Users@openser.org">Users@openser.org</a><br><a href="http://openser.org/cgi-bin/mailman/listinfo/users">http://openser.org/cgi-bin/mailman/listinfo/users </a><br></blockquote></div><br>
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