There is definitely wrong in your setup . I have ipkall setup on my asterisk and dont have ports 1000-2000 open ( only 10000-20000,5060,4569 open ) . and incoming calls word fine for me .<br><br><div><span class="gmail_quote">
On 14/11/06, <b class="gmail_sendername">Al Bochter</b> <<a href="mailto:Al.Bochter@bochterservices.com">Al.Bochter@bochterservices.com</a>> wrote:</span><blockquote class="gmail_quote" style="margin-top: 0; margin-right: 0; margin-bottom: 0; margin-left: 0; margin-left: 0.80ex; border-left-color: #cccccc; border-left-width: 1px; border-left-style: solid; padding-left: 1ex">
No 1000 to 2000 is not a typo.<br>Well let me put some light on this......<br><br>If you goto <a href="http://www.ipkall.com/">http://www.ipkall.com/</a><br>and your firewall is set to 10000 to 20000 you WILL NOT get SIP calls
<br>from <a href="http://www.ipkall.com/">http://www.ipkall.com/</a> DID's<br><br>As soon as you OPEN ports 1000 to 2000 to the PBX Server the calls from<br><a href="http://www.ipkall.com/">http://www.ipkall.com/</a> will work fine.
<br><br>You DON'T have to make any changes to /etc/asterisk/rtp.conf<br><br>This is what I ran into today<br><br>So I guess you are right... It's a free for all on ports. Makes things<br>harder to do.<br>I think we need to get a better standard just to make this easier.
<br><br>// There's no standard - there are several different conventions adopted<br>// by different vendors, though.<br><br>Best regards,<br><br>Al Bochter<br>Bochter Services<br><a href="http://www.BochterServices.com/?t=Email">
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<br><br><br>Peter Bowyer wrote:<br><br>> On 13/11/06, Al Bochter <<a href="mailto:Al.Bochter@bochterservices.com">Al.Bochter@bochterservices.com</a>> wrote:<br>><br>>> Yes you are right 10000-20000 are rtp ports used by asterisk by default
<br>>> I have some that do set a custom range in /etc/asterisk/rtp.conf ..<br>>><br>>> After looking around.. There were not any notes about the 1000 - 2000<br>>> port<br>>> range on there website.
<br>>> As you know if you don't know what the ports are it no workie!!!!!<br>>> And it is not good to DMZ the server.....<br>>> ----------<br>>> Now I have a handytone 386 that is set to<br>>>
<br>>> SIP port 5060 and 5062<br>>> RTP port 5004 and 5008<br>>><br>>> You can set Random Ports to use: 1024 to 65535<br>>><br>>> The handytone will work fine on the LAN.... But if you would moved the
<br>>> Handytone to the internet it would NOT work do to the firewall..<br>>> Using the asterisk defaults<br>>> ----------<br>>> So liked I ask before "So is there any standard ports"<br>
><br>><br>> Both sides have to be willing to negotiate a port. Maybe your<br>> handytone has its own restrictions on RTP ports? As you now know,<br>> Asterisk doesn't care as long as you specify a range in
rtp.conf.<br>><br>> 1000-2000 must be a typo as ports <1024 are reserved and privileged.<br>><br>> There's no standard - there are several different conventions adopted<br>> by different vendors, though.
<br>><br>> <a href="http://en.wikipedia.org/wiki/Real-time_Transport_Protocol">http://en.wikipedia.org/wiki/Real-time_Transport_Protocol</a> might help.<br>><br>> Peter<br><br>_______________________________________________
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