Elpidio,
Glad to hear it. Depending on your config, you may need to allow the
RTP ports through as well. I poked holes in my firewall for ports
10000-20000, probably overkill, though.
Bob...
Elpidio Ramos wrote:
This helped a lot.
It was the firewall. I got it configured right now.
Thanks
*/Bob Chiodini <[EMAIL PROTECTED]>/* wrote:
I think all anywhere should allow 5060. Try running service iptables
stop (as root) to shutdown the firewall. See if 5060 then answers.
I'm not running a firewall on my asterisk box so I'm not sure what
the
rule would need to be. service iptables start will restore the
firewall.
Bob...
Elpidio Ramos wrote:
> Bob,
>
> I get the same answer you get when using netstat -an
>
> When I query the firewall rules I get this:
> Chain RH-Firewall-1-INPUT (2 references)
> target prot opt source destination
> ACCEPT all -- anywhere anywhere
> ACCEPT icmp -- anywhere anywhere icmp any
> ACCEPT ipv6-crypt-- anywhere anywhere
> ACCEPT ipv6-auth-- anywhere anywhere
> ACCEPT udp -- anywhere 224.0.0.251 udp dpt:5353
> ACCEPT udp -- anywhere anywhere udp dpt:ipp
> ACCEPT all -- anywhere anywhere state
> RELATED,ESTABLISHED
> ACCEPT tcp -- anywhere anywhere state NEW
> tcp dpt:ssh
> ACCEPT tcp -- anywhere anywhere state NEW
> tcp dpt:http
> REJECT all -- anywhere anywhere
> reject-with icmp-host-prohibited
>
> I assume this indicates port 5060 is restricted?
>
> Elpidio
>
> */Bob Chiodini /* wrote:
>
> Elpidio,
>
> Is it truly not listening or is maybe a firewall blocking port 5060.
> What does netstat -an | grep 5060 tell you? I get this:
>
> netstat -an | grep 5060
> udp 0 0 0.0.0.0:5060 0.0.0.0:*
>
> iptables -L will list any firewall restrictions.
>
> Bob...
>
>
> On Fri, 2006-09-01 at 09:12 -0700, Elpidio Ramos wrote:
> > Does anyone knows what could be the cause for asterisk not
listening
> > in post 5060 if SIP interfaces is loaded with no problems?
> >
> > I am using Fedora Core 3.
> >
> > I have followed the instructions in several tutorials and tried
> > several soft phones and the SIP interface seem to be dead.
> >
> > 1. When loading asterisk SIP load with no problem
> > 2. When I activate the DEBUG for a peer, ip or sip in general, I
> don't
> > get to see any messages when a connection is attempted from
any soft
> > phone.
> > 3. The soft phones all report a timeout when trying to register.
> > 4. Tried also to move to port 80 that I know is open but still the
> > same problem.
> >
> > I will appreciate any help anyone can provide with this problem.
> >
> > Elpidio
> > _______________________________________________
> > --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
>
>
>
>
> *Elpidio Ramos*
> President
> RM International Services SA CV
> Web: http://www.ramosoft.com
> Mex: +52 (55) 5116-9804 Office
> +52 (55) 5116-9805 Fax
> +52 (55) 1755-6601 Cell
> USA: +1 (801) 494-1415 Office
> +1 (240) 250-8264 Fax
> +1 (801) 938-4740 Direct
>
*Elpidio Ramos*
President
RM International Services SA CV
Web: http://www.ramosoft.com
Mex: +52 (55) 5116-9804 Office
+52 (55) 5116-9805 Fax
+52 (55) 1755-6601 Cell
USA: +1 (801) 494-1415 Office
+1 (240) 250-8264 Fax
+1 (801) 938-4740 Direct
_______________________________________________
--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
