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<BR><FONT face="Verdana, Geneva, Arial, Sans-serif">no.. it's directly connected to the internet.. it's not an issue of accepting calls.. see.. the problem is the call gets to the server.. the server tries to route it.. </FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">but as if the dial plan is not there.. it rejects the call because it doesn't know what to do with it.. </FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">for example of my SIP.Conf</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif"></FONT> <BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">[5003]</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">type=peer</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">qualify=yes</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">port=5060</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">nat=yes</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">host=HOSTIP</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">allow=all</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">dial=SIP/5003</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">context=from-smarttech</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">canreinvite=no</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">call-limit=50</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">deny=0.0.0.0/0.0.0.0</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">permit=HOSTIP/255.255.255.255</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif"></FONT> <BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">Extensions.conf</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">[from-smarttech]</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">exten => fax,1,Goto(ext-fax,in_fax,1)</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">exten => s,n,Set(__FROM_DID=${EXTEN})</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">exten => s,n,Gosub(app-blacklist-check,s,1)</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">exten => s,n,GotoIf($[ "${CALLERID(name)}" != "" ] ?cidok)</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">exten => s,n,Set(CALLERID(name)=${CALLERID(num)})</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">exten => s,n(cidok),Noop(CallerID is ${CALLERID(all)})</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">exten => s,n,Set(__CALLINGPRES_SV=${CALLINGPRES_${CALLINGPRES}})</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">exten => s,n,SetCallerPres(allowed_not_screened)</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">exten => s,n,Goto(ext-queues,8004,1)</FONT><BR>
<BR><FONT face="Verdana, Geneva, Arial, Sans-serif">let's say smarttech is a voip provider.. which forwards calls to my user on their system .. now my server is supposed to route those calls according to the dial plan.. </FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">the same exact settings worked like magic on another server.. but on this server.. it just as if the context and the dial plan does not exist.!!!</FONT><BR>
<FONT face="Verdana, Geneva, Arial, Sans-serif">any idea?</FONT><BR><FONT color=#808080><STRONG>
<HR id=EC_[object]>
<BR>
<BR><BR></STRONG></FONT>
<P align=left><FONT color=#808080><STRONG>AHD Tarek Sawah</STRONG></FONT></P>
<P align=left><FONT color=#808080><EM>Integrated Digital Systems</EM></FONT></P>
<P align=left><FONT color=#808080><EM>CCNA, MCSE, RHCE, VoIP</EM></FONT></P>
<P align=left><FONT color=#808080><EM>Syria: +963 944 618286</EM></FONT></P>
<P align=left><FONT color=#808080><EM>USA: +1 347 562 2308</EM></FONT></P><A href="http://www.tareksawah.com/" target=_blank></A><BR><BR>
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Date: Fri, 26 Sep 2008 11:55:45 -0500<BR>From: brent@texascountrytitle.com<BR>To: asterisk-users@lists.digium.com<BR>Subject: Re: [asterisk-users] Dial Plan Issues<BR><BR>
<META content="Microsoft SafeHTML" name=Generator>Steve Murphy wrote:
<BLOCKQUOTE id=EC_mid_1222353300_24346_19_camel_digium2 cite=mid:1222353300.24346.19.camel@digium2><PRE>On Thu, 2008-09-25 at 14:21 +0000, Tariq .. wrote:
</PRE>
<BLOCKQUOTE id=EC_StationeryCiteGenerated_1><PRE>Greetings,
i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..
i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction..
i have the same exact settings for the extensions.conf
i tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls..
so my question is..
is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on..
what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other..
regards
________________________________
</PRE></BLOCKQUOTE><PRE>Tariq--
You might try a trixbox users mailing list.
There might be a few trixbox users hanging around in
this group who might be able to help, but your
chances are much better in that list.
murf
</PRE><PRE></PRE></BLOCKQUOTE><BR>The server that is not accepting calls is not behind a NAT firewall by any chance is it?<BR><br /><hr />Stay up to date on your PC, the Web, and your mobile phone with Windows Live. <a href='http://clk.atdmt.com/MRT/go/msnnkwxp1020093185mrt/direct/01/' target='_new'>See Now</a></body>
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