[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}@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.
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From: <i>Mark Aiken <aiken.mark at gmail.com></i><br>Reply-To: <i>Mark
Aiken <aiken.mark at gmail.com></i><br>To: <i>Iqbal
<iqbal at gigo.co.uk></i><br>CC: <i>Bogdan-Andrei Iancu
<bogdan at voice-system.ro>, "Matt L. Zhu"
<coder0000 at hotmail.com>, 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>
<br>
Mark<br><br><div><span class="gmail_quote">On 9/29/05, <b
class="gmail_sendername">Iqbal</b> <<a
href="mailto:iqbal at gigo.co.uk">iqbal at 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>>>
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list<br>>>
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