<DIV>there is no error message coming up on the pbx for in-bound calls (there is only debugging messages for outbound calls).</DIV> <DIV> </DIV> <DIV>thanks in advance for any hint or suggestion.</DIV> <DIV>Ama</DIV> <DIV> </DIV> <DIV>I just post my configuration file here for sip phone:</DIV> <DIV>extensions.conf<BR>-------------------------------------------------------------------------<BR>[globals]</DIV> <DIV>[default]<BR>include => incoming<BR>include => outgoing<BR>include => iax<BR>inculde => sip<BR>include => sccp<BR>[sip]<BR>exten => 2171,1,Dial(SIP/stargate1,20)<BR>;exten => 2171,1,Dial(SIP/2171,20)<BR>exten => 2171,2,Hangup<BR>exten => 2172,1,Dial(SIP/stargate2,20)<BR>;exten => 2172,1,Dial(SIP/2172,20)<BR>exten => 2172,2,Hangup<BR>exten => 2173,1,Dial(SIP/stargate3,20)<BR>;exten => 2173,1,Dial(SIP/2173,20)<BR>exten => 2173,2,Hangup</DIV> <DIV>[sccp]</DIV> <DIV>[skinny]</DIV> <DIV>[incoming]<BR>exten =>
_214943[5-9]6,1,Dial(SIP/stargate3)<BR>exten => _214943[5-9]6,2,Hangup</DIV> <DIV>[outgoing]<BR>exten => _XXXXXXXX,1,Dial(Zap/g1/${EXTEN})<BR>exten => _XXXXXXXX,2,Hangup<BR>-------------------------------------------------------------------------</DIV> <DIV>sip.conf<BR>-------------------------------------------------------------------------<BR>[general]<BR>context=default ; Default context for incoming calls<BR> ; Set this to your host name or domain name<BR>bindport=5060 ; UDP Port to bind to (SIP standard port is
5060)<BR>bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)<BR>srvlookup=yes ; Enable DNS SRV lookups on outbound calls<BR> </DIV> <DIV>register => stargate1:1stargate@local_sip/2171<BR>register => stargate2:2stargate@local_sip/2172<BR>register => stargate3:3stargate@local_sip/2173<BR>;---------------------------------------------- NAT SUPPORT ------------------------<BR>nat=no ; Global NAT settings (Affects all peers and
users)<BR> </DIV> <DIV><BR>[local_sip]<BR>type=friend<BR>host=10.47.200.136<BR>context=default</DIV> <DIV>[stargate1] ;cisco 9760<BR>;[2171]<BR>type=friend<BR>host=dynamic ;10.47.200.140 ;dynamic<BR>defaultip=10.47.200.140<BR>username=stargate1<BR>secret=xxx<BR>callerid="21495071" <2171><BR>allow=all<BR>qualify=200<BR>nat=no<BR>defaultip=10.47.200.140</DIV> <DIV><BR>[stargate2] ;Polycom 601<BR>;[2172]<BR>type=friend<BR>host=dynamic ;10.47.200.141 ;dynamic<BR>defaultip=10.47.200.141<BR>username=xxx<BR>secret=2stargate<BR>callerid="21495072" <2172><BR>allow=all<BR>qualify=200<BR>nat=no<BR>defaultip=10.47.200.141</DIV> <DIV>[stargate3] ;Aastra 480i<BR>;[2173]<BR>type=friend<BR>host=dynamic ;10.47.200.137 ;dynamic<BR>defaultip=10.47.200.137<BR>username=stargate3<BR>callerid="stargate3"
<2173><BR>secret=xxx<BR>allow=all<BR>qualify=200<BR>nat=no<BR>defaultip=10.47.200.137<BR>----------------------------------------------------------------------------<BR></DIV> <DIV><BR><B><I>pdhales@optusnet.com.au</I></B> wrote:</DIV> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid"> <META content="MSHTML 6.00.2800.1528" name=GENERATOR> <STYLE></STYLE> <DIV><FONT face=Arial size=2>What error do you get when trying to call the SIP phones?</FONT></DIV> <DIV><FONT face=Arial size=2></FONT> </DIV> <DIV><FONT face=Arial size=2>PaulH</FONT></DIV> <DIV> </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 11:58 PM</DIV> <DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] Help with sip setup because can't receive calls</DIV> <DIV><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></BLOCKQUOTE>_______________________________________________<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></BLOCKQUOTE> <DIV><BR></DIV><p>
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