Hi Elpidio,<br><br>I am Chandra from India. I have a doubt. I am trying to solve my problem from many days. But, I couldn't able to solve this problem. I am using Asterisk with Redhat Enterprise Linux 4.2. The problem is 5060 port is blocked. After stop my firewall (service iptables stop) also, 5060 port is not opening. I checked with the below command:<br># nmap -p5060 192.168.91.22--->This is my IP address<br>and it is showing that port 5060 is closed. How can I enable and open this 5060 port? Really, I am breaking my head with this problem. SIP is not working because of this problem. Please tell me a solution. Looking forward to your reply. Thank you.<br><br>Regards,<br>Chandra.<br><br><br><br><br><b><i>Elpidio Ramos <elpidio@ramosoft.com></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> <div>Hi,</div> <div> </div> <div>This is a sample file I am currently using on my
server.</div> <div>My server has a public IP address and an internal IP address (duan NIC).</div> <div>It runs Fedora Core 3 running iptables firewall already configured with ports </div> <div>4569, 5060, 10000-20000 open (udp and tcp)</div> <div> </div> <div>[general]<br>context=default<br>allowguest=no<br>realm=your.hostname.ext<br>bindaddr=0.0.0.0<br>bindport=5060<br>externip=your.server.ip.address<br>srvlookup=no<br>maxexpirey=3600<br>disallow=all<br>allow=ulaw<br>allow=ilbc<br>allow=gsm<br>musicclass=default<br>language=es<br>rtptimeout=120<br>rtpholdtimeout=300<br>useragent=asterisk<br>localnet=10.10.10.0/255.255.255.0<br>rtcachefriends=no<br>qualify=yes</div> <div> </div> <div>[311]<br>type=friend<br>regexten=311<br>username=311<br>secret=311<br>callerid="User on extension 311" <311><br>host=dynamic<br>nat=yes<br>canreinvite=no</div> <div> </div> <div>[312] <br>type=friend<br>regexten=312<br>username=312<br>secret=312<br>callerid="User
on extension 312" <312><br>host=dynamic<br>nat=yes<br>canreinvite=no</div> <div><br><br><b><i>tengulre <tengulre@megamail.com.cn></i></b> wrote:</div> <blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); padding-left: 5px; margin-left: 5px;"> <meta content="MSHTML 6.00.2900.2963" name="GENERATOR"> <style></style> <div><font size="2"> How to using SIP to connect remote other VoIP server? is it only running one line voice if I registered a one SIP account?</font></div> <div><font size="2"> anybody can give me some sample configuration files? thanks a lot!</font></div> <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><br><br><div id="RTEContent"> <div id="RTEContent">
<div id="RTEContent"> <div> <div> <div> <div> <div> <div> <div><strong><font size="3">Elpidio Ramos</font></strong> <br>President <br>RM International Services SA CV <br>Web: http://www.ramosoft.com <br>Mex: +52 (55) 5116-9804 Office<br> +52 (55) 5116-9805 Fax </div> <div> +52 (55) 1755-6601 Cell<br>USA: +1 (801) 494-1415 Office</div> <div> +1 (240) 250-8264 Fax</div> <div> +1 (801) 938-4740 Direct</div> <div> </div></div></div></div></div></div></div></div></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><p> 
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