<DIV>Hi </DIV> <DIV>I basically allow=all and NAT=no for all the phones. but still can't see why I can't receive calls (i.e. in-bound) but I can make outbound calls. also there is no debuging on pbx for sip (unless it's outbound call).</DIV> <DIV>do you have anymore advice?</DIV> <DIV> </DIV> <DIV>thanks</DIV> <DIV>Ama<BR><BR><B><I>Md Sani Johari <sanijohari@trisys.com.my></I></B> wrote:</DIV> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid"> <META content="MSHTML 6.00.2900.2180" name=GENERATOR> <STYLE></STYLE> <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> <div> <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> <div> <HR> <div></div>_______________________________________________<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> <div> <HR> <div></div>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:
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