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Where is your DMZ pointed?<br>
<pre class="moz-signature" cols="72">Best regards,
Al Bochter
Bochter Services
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<br>
<br>
Vicky wrote:
<blockquote
cite="midff95c3730611132328mea6c524j533dad0e4b51af31@mail.gmail.com"
type="cite">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="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; 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">http://www.BochterServices.com/?t=Email</a><br>
<br>
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<br>
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<br>
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<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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Inbound (clean). Database: 0648-1, 11/13/2006 - 11/14/2006 2:29:51 AM
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