I have shifted asterisk port to 5091 . Now i am able to register properly using sjphone but still when dialing number it keep on showing calling .. and do not go ahead . I change asterisk's rtp ports too but still i am unable to make call . My other softphone on different internet isp is working properly . :(
<br><br><div><span class="gmail_quote">On 16/12/06, <b class="gmail_sendername">Tim C. Lewis</b> <<a href="mailto:tclewis@oreilly.com">tclewis@oreilly.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>well that should map incoming packets to 5091 to 5060, but may not rewrite<br>[new] outbound packets from 5060 to 5091, which your isp may be blocking.<br>an iptables SNAT or MASQUERADE might help you there. i'm not positive on
<br>if this would be needed or not.<br><br>more importantly, however, if your isp is blocking all outgoing traffic to<br>5060, it won't get to your softphone anyway, unless you also configure<br>that end to also not use 5060. and if you're reconfiguring ports on the
<br>softphone end anyway, why not just put 5091 in there, 5091 in sip.conf's<br>bindport, and not mess with iptables at all?<br><br>another option might be that your isp is blocking rtp as well.<br><br>can you see what the asterisk console is doing when you attempt such
<br>calls? and/or tcpdump?<br><br>-tcl.<br><br><br>On Sat, 16 Dec 2006, Mail list wrote:<br><br>> Hello my isp has blocked outgoing and incoming connection for port 5060 . I<br>> have ssh access to server so i want to send all traffic from port 5091 to
<br>> port 5060 of asterisk .so i tried<br>><br>> iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to<br>> <a href="http://127.0.0.1:5060">127.0.0.1:5060</a><br>><br>> Now my softphone is able to register with asterisk but it isnt able to make
<br>> any calls .<br>><br>> bindport = 5091 in my sip.conf under extensions is not working .. asterisk<br>> doesnt listen to port 5091 .. but if i put in general section of<br>> sip.confthen it works but then asterisk wont listen on 5060 . How can
<br>> i use iptables<br>> in this situation ?<br>><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list
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