[asterisk-users] SIP Ports (1000 to 2000 works)

[Al Bochter]

No 1000 to 2000 is not a typo.
Well let me put some light on this......

If you goto http://www.ipkall.com/
and your firewall is set to 10000 to 20000 you WILL NOT get SIP calls 
from http://www.ipkall.com/ DID's

As soon as you OPEN ports 1000 to 2000 to the PBX Server the calls from 
http://www.ipkall.com/ will work fine.

You DON'T have to make any changes to /etc/asterisk/rtp.conf

This is what I ran into today

So I guess you are right... It's a free for all on ports. Makes things 
harder to do.
I think we need to get a better standard just to make this easier.

// There's no standard - there are several different conventions adopted
// by different vendors, though.

Best regards,

Al Bochter
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Peter Bowyer wrote:

> On 13/11/06, Al Bochter <Al.Bochter at bochterservices.com> wrote:
>
>> Yes you are right 10000-20000 are rtp ports used by asterisk by default
>> I have some that do set a custom range in /etc/asterisk/rtp.conf ..
>>
>> After looking around.. There were not any notes about the 1000 - 2000 
>> port
>> range on there website.
>> As you know if you don't know what the ports are it no workie!!!!!
>> And it is not good to DMZ the server.....
>> ----------
>> Now I have a handytone 386 that is set to
>>
>> SIP port 5060 and 5062
>> RTP port 5004 and 5008
>>
>> You can set Random Ports to use:  1024 to 65535
>>
>> The handytone will work fine on the LAN.... But if you would moved the
>> Handytone to the internet it would NOT work do to the firewall..
>> Using the asterisk defaults
>> ----------
>> So liked I ask before  "So is there any standard ports"
>
>
> Both sides have to be willing to negotiate a port. Maybe your
> handytone has its own restrictions on RTP ports? As you now know,
> Asterisk doesn't care as long as you specify a range in rtp.conf.
>
> 1000-2000 must be a typo as ports <1024 are reserved and privileged.
>
> There's no standard - there are several different conventions adopted
> by different vendors, though.
>
> http://en.wikipedia.org/wiki/Real-time_Transport_Protocol might help.
>
> Peter