<html>
<head>
<style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Tahoma
}
--></style>
</head>
<body class='hmmessage'>
Ok. Problem solved . <div><br></div><div>Thank you very much!!!<br><br>Att,<br>
<br>
Flavio Roberto Miranda<br>
MSN:flaviormiranda@hotmail.com<br>Skype: flaviormiranda<br><br><br><br><hr id="stopSpelling">Date: Wed, 15 Sep 2010 09:56:36 -0400<br>From: zishanov@gmail.com<br>To: kpfleming@digium.com; asterisk-users@lists.digium.com<br>Subject: Re: [asterisk-users] incoming call FXO<br><br>As Kevin said, you need to define an 's' extension where the calls will be answered. Seems like you are using default configuration. Open file 'extensions.conf' in /etc/asterisk folder and look for context named [default]. If it is not there, create one and add something under it, e.g.,<BR>
[default]<br>
exten => s,1,Verbose( - - - Call received - - - )<br>
exten => s,n,Playback(hello-world)<br>
extent => s,n,HangUp()<BR>
Then do a 'core extensions reload' on Asterisk CLI. Now calling in on FXO should play the message 'hello-world' (assuming this sound file exists in the sound folder of asterisk), and you'll see the call activity on the CLI.<BR>
For the rest, you need to consult chapters 5 and 6 of 'Asterisk-The Future of Telephony' book.<BR>
Zeeshan A Zakaria<BR>
--<br>
<a href="http://www.ilovetovoip.com" target="_blank">www.ilovetovoip.com</a><BR>
<BR><blockquote>On 2010-09-15 8:59 AM, "Kevin P. Fleming" <<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>> wrote:<br><br><font color="#500050">On 09/15/2010 07:20 AM, Flavio Miranda wrote:<br>
<br>> Recently I have instaled one Digium TDM410 on my...</font><BR>Right, that's what the message is telling you. For incoming calls on<br>
FXO, they can *only* be sent to the 's' extension in the target context,<br>
since there is no target number passed over the FXO connection. You'll<br>
have to create an 's' extension to handle incoming calls however you like.<br>
<br>
--<br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
skype: kpfleming | jabber: <a href="mailto:kfleming@digium.com">kfleming@digium.com</a><br>
Check us out at <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote><BR>
<br>--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users</div> </body>
</html>