<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2900.2180" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT face=Arial size=2>hi abc def,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>what type of voice codec that phone use. Maybe it
can't support. </FONT></DIV>
<DIV><FONT face=Arial size=2>I also have same problem my sip phone, when i
change the voice codec from </FONT><FONT face=Arial size=2>
g729 to g711 ulaw, then it work find.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>also make sure wether your sip is behind the router
or not..</FONT></DIV>
<DIV><FONT face=Arial size=2>nat=never</FONT></DIV>
<DIV><FONT face=Arial size=2>or</FONT></DIV>
<DIV><FONT face=Arial size=2>nat=1</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=xisterisk@yahoo.com href="mailto:xisterisk@yahoo.com">abc def</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Wednesday, January 25, 2006 8:58
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] Help with sip
setup because can't receive calls</DIV>
<DIV><FONT face=Arial size=2></FONT><FONT face=Arial size=2></FONT><BR></DIV>
<DIV id=RTEContent>
<DIV id=RTEContent>Hi all,</DIV>
<DIV>I read many posts on asterisk mail site and been trying many
different things but still I can't get my sip phones to work with
asterisk.<BR> I have a full blown-up voip netwok with two asterisk
servers connected <BR>to pstn network with iax phones and cisco sccp
phones which all work fine. <BR>however, I have been struggeling to configure
my sip phones (polycom 601, Aastra 480i and cisco 9760) to work with asterisk.
I can call out from sip phones to anywhere else but not receive phone calls. I
can see the phones on "sip show registry" and "sip show peers" but no track
phone calls for sip.<BR> <BR> can you please shed some light
on me how to go about solving this <BR>problem?<BR> <BR>
thank you and best regards,<BR> Ama<BR></DIV></DIV>
<P>
<HR SIZE=1>
Do you Yahoo!?<BR>With a free 1 GB, there's more in store with <A
href="http://us.rd.yahoo.com/mail_us/taglines/mailstorage/*http://mail.yahoo.com/">Yahoo!
Mail.</A>
<P>
<HR>
<P></P>_______________________________________________<BR>--Bandwidth and
Colocation provided by Easynews.com --<BR><BR>Asterisk-Users mailing
list<BR>To UNSUBSCRIBE or update options visit:<BR>
http://lists.digium.com/mailman/listinfo/asterisk-users<BR>
<P>
<HR>
<P></P>No virus found in this incoming message.<BR>Checked by AVG Free
Edition.<BR>Version: 7.1.375 / Virus Database: 267.14.22/238 - Release Date:
23/01/2006<BR></BLOCKQUOTE></BODY></HTML>