<div>Hi,</div>
<div> </div>
<div>Check the codec used on your sip.conf </div>
<div> </div>
<div>make allow-ulaw</div>
<div> </div>
<div>It will work fine, i had the sae problem cause i was using Ilbc.</div>
<div> </div>
<div>Regards,<br><br> </div>
<div><span class="gmail_quote">On 5/22/07, <b class="gmail_sendername">Milton Davila</b> <<a href="mailto:davila.milton@gmail.com">davila.milton@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"><span>Hi there, <br><br>I am having some problems while trying to place phone calls through Asterisk to Net2phone, this is my setup:
<br><br>I have a SIP phone connected directly to my Asterisk box from where I want the call to origin; in sip.conf: <br><br>[mySIP] <br>type=friend <br>username=mySIP <br>secret=mySecret <br>host=dynamic <br>context=outgoing
<br><br>I read that I have to make some changes in sip.conf, in order to make it work with Net2phone: <br><br><a onclick="return top.js.OpenExtLink(window,event,this)" href="http://www.voip-info.org/wiki/view/Asterisk+settings+Net2phone" target="_blank">
http://www.voip-info.org/wiki/view/Asterisk+settings+Net2phone</a> <br><br>So these are the changes I made in sip.conf: <br><br>[general] <br>useragent = X-Lite release 1103m <br>register => <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:PHONENUMBER:PASSWORD@sip.net2phone.com" target="_blank">
PHONENUMBER:PASSWORD@sip.net2phone.com</a> <br><br>[net2phone] <br>type = peer <br>host = <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://sip.net2phone.com/" target="_blank">sip.net2phone.com</a> <br>
username = PHONENUMBER <br>secret = PASSWORD <br>fromuser = PHONENUMBER <br>fromdomain = <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://net2phone.com/" target="_blank">net2phone.com</a> <br>context = incoming
<br>insecure = very <br>canreinvite = no <br><br>Now here is my extensions.conf: <br><br>[outgoing] <br>exten => _9NXXNXXXXXX,1,Dial(SIP/net2phone/${EXTEN:1}) <br><br>If I type "sip show registry" in the Asterisk console, it shows that the state of the Net2phone sip is "Registered".
<br><br>The problem is that when I call any phone is USA: 1-XXX-XXX-XXXX, I only get a busy tone. So I can never really place a call. <br><br>What can be the problem? <br><br>I am using Asterisk 1.4.2 on Red Hat Enterprise Linux 5.
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