<DIV>Hi all, I am resending this message, so far no one has helped me with this incoming call issue. there is no problem with outbound call but there is no inbound call to my sip phone. the only message I get when I call from pstn is "unable to create local channel for call forward to <A href="mailto:'Local/sipphone@default'">'Local/sipphone@default'</A> (case =0)". my configuration files are attached below. any help would be greatly appreciated. many thanks in advance.<BR>ABC</DIV> <DIV><BR><B><I>abc def <xisterisk@yahoo.com></I></B> wrote:</DIV> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid"> <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="starg ate3" <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></DIV></DIV></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> <div> <HR SIZE=1> Bring words and photos together (easily) with<BR><A href="http://us.rd.yahoo.com/mail_us/taglines/PMHM3/*http://photomail.mail.yahoo.com">PhotoMail </A>- it's free and works with Yahoo!
Mail._______________________________________________<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>
<hr size=1> <a href="http://us.rd.yahoo.com/evt=38381/ ylc=X3oDMTEzcGlrdGY5BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDMWF1dG9z/*http://autos.yahoo.com/index.html ">Yahoo! Autos</a>. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars.