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

[Al Bochter]

Where is your DMZ pointed?

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

Are you outside of the US?
Do you need to call US Toll Free Numbers?
We can help you save money on calling US toll free numbers.

Email for information: usTollFree at bochterservices.com

(Cellular) 1-712-432-5401

(Voip PBX) Free World DialUp: 780-217 EXT: 250
WebSite: http://www.freeworlddialup.com/

BUY and sell Coins, Silver and Gold
http://www.bochterservices.com/?j=gold&t=email

For new and used security items
http://www.bochterservices.com/?j=store&t=email_security

GOLD PLATING SERVICES
http://www.bochterservices.com/?j=plating&t=email



Vicky wrote:

> 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 .
>
> On 14/11/06, *Al Bochter* <Al.Bochter at bochterservices.com 
> <mailto:Al.Bochter at bochterservices.com>> wrote:
>
>     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
>     Bochter Services
>     http://www.BochterServices.com/?t=Email
>
>     Are you outside of the US?
>     Do you need to call US Toll Free Numbers?
>     We can help you save money on calling US toll free numbers.
>
>     Email for information: usTollFree at bochterservices.com
>     <mailto:usTollFree at bochterservices.com>
>
>     (Cellular) 1-712-432-5401
>
>     (Voip PBX) Free World DialUp: 780-217 EXT: 250
>     WebSite: http://www.freeworlddialup.com/
>
>     BUY and sell Coins, Silver and Gold
>     http://www.bochterservices.com/?j=gold&t=email
>     <http://www.bochterservices.com/?j=gold&t=email>
>
>     For new and used security items
>     http://www.bochterservices.com/?j=store&t=email_security
>     <http://www.bochterservices.com/?j=store&t=email_security>
>
>     GOLD PLATING SERVICES
>     http://www.bochterservices.com/?j=plating&t=email
>     <http://www.bochterservices.com/?j=plating&t=email>
>
>
>
>     Peter Bowyer wrote:
>
>     > On 13/11/06, Al Bochter <Al.Bochter at bochterservices.com
>     <mailto: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
>
>     _______________________________________________
>     --Bandwidth and Colocation provided by Easynews.com
>     <http://Easynews.com> --
>
>     asterisk-users mailing list
>     To UNSUBSCRIBE or update options visit:
>        http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>------------------------------------------------------------------------
>
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>asterisk-users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>
>------------------------------------------------------------------------
>
>
>
>----------------------------------------------------
>Inbound (clean). Database: 0648-1, 11/13/2006 - 11/14/2006 2:29:51 AM
>
>
>
>
>  
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061114/01c72cc5/attachment.htm