Hi,
I think that the web-driven SIP Phone (free) doddle (beta version) can be
useful with your Asterisk applications.
You can pre-fill it with your sip settings (Asterisk host name or IP / realm /
sip user), you just need to setup the HTML link as that: (Attached is the HTML
page example)
/**************************/
simple HTML code example:
/*************************/
<html>
<head>
<script type="text/javascript">
function webcall_win(sip,realm,phone,user,serviceName)
{
//You can have your ajax code here communicating with your site...
//XMLHttpRequest...
var URL =
"http://doddle.com.br/endoddle.jsp?sipserver="+sip+"&siprealm="+realm+"&callto="+phone+"&username="+user+"&provider="+serviceName;
window.open(URL,"MyWindow")
}
</script>
</head>
<body>
<h3>Your Asterisk Applications web site...</h3>
<p>Use Asterisk to call right now!
<a
href="javascript:webcall_win('asteriskIP','asterisk','123456','myuser','myServiceName')";"><u>Web-driven
Call</u></a>
</body>
</html>
/*********************************/
Thus your Asterisk sip users are ready to call from web page with your Asterisk
server.
PS: Asterisk’s default realm: asterisk
sip.conf:
[general]
realm = your_realm_here / default is asterisk
Address: www.doddle.com.br
Paulo
My Asterisk Applications web site here...
Use our Asterisk to call right now! Web-driven Call
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