[asterisk-users] SIP Ports (1000 to 2000 works)
Vicky
vicky.r at gmail.com
Mon Nov 13 15:52:05 MST 2006
FRom voip-info.org
# SIP on UDP port 5060. Other SIP servers may need TCP port 5060 as well
iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT
# IAX2- the IAX protocol
iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT
# IAX - most have switched to IAX v2, or ought to
iptables -A INPUT -p udp -m udp --dport 5036 -j ACCEPT
# RTP - the media stream
iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT
# MGCP - if you use media gateway control protocol in your configuration
iptables -A INPUT -p udp -m udp --dport 2727 -j ACCEPT
Open all above ports and you should be good to go . Maybe you are recieving
calls over iax and u havent opened iax2 port 4569 .. Anyway my server has
all above ports opened and i have zero problems :) .
On 14/11/06, 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"
>
> 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:
>
> actually 10000-20000 are rtp ports used by asterisk .. its not really
> compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check
> ur rtp.conf what range its using and open that in firewall . Default with
> asterisk is 10000-20000 unless changed .
>
> On 14/11/06, Al Bochter <Al.Bochter at bochterservices.com> wrote:
> >
> > I was reading the posts and someone said about the default 1000 to 2000
> > I see in the .conf the default is 10000 to 20000
> >
> > I found a service that gives inbound DID's in the firewall 5060 and
> > 10000 - 20000 is setup
> > no workie on the DID
> >
> > But when I set 5060 , 10000 - 20000 and (Unblocked) 1000 - 2000
> > Now the DID works fine.
> >
> > So you me what the standard is
> >
> > --
> > 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
> >
> > _______________________________________________
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