But that's the problem, I didn't. Not until you suggested it later on.
While I originally had port 5060 configured in the plugin and being
used on the ATA, when that didn't work I changed the port on the ATA to
5061 and later on 5090. Both times without modifying the voip-sip
plug-in at all. An
If you had previously put 5090 into SIP_VOIP_PORTS then yes, that would
have persisted across firewall restarts.
Hence the need to reboot.
On 01/25/2010 09:33 AM, James Babiak wrote:
> Hey Everyone,
>
> Ok, so I think I got everything working. It was the voip-sip plugin
> that was causing the pr
Hey Everyone,
Ok, so I think I got everything working. It was the voip-sip plugin that was
causing the problem. I had to disable it altogether and then reboot the
astlinux box. Restarting only the firewall/iptables had no affect. It seems
like the plugin is broken, because if enabled, it will appa
Trying adding 5090 to the port list, and reboot.
And yes, nf_conntrack_sip and nf_nat_sip *will* rewrite INVITE's.
Though usually only outbound. There's no reason to inbound.
On 01/24/2010 07:40 PM, Lonnie Abelbeck wrote:
> James,
>
> I also have a SPA-3102 (voice, no FAX) behind NAT, behind A
James,
I also have a SPA-3102 (voice, no FAX) behind NAT, behind AstLinux 0.7
---
SPA-3102 [SIP]
NAT Support Parameters
Handle VIA received: yesHandle VIA rport: yes
Insert VIA received: yesInsert VIA rport: yes
Substitute VIA Addr: no Send Resp To Src Port
Hey,
Thanks for the assistance everyone.
.
The reason why I left 5090 out of the firewall's SIP plugin was because
I am port forwarding 5090 directly to the ATA to keep Asterisk out of
the mix. When I initially began testing this, before I made any changes
on the ATA or Astlinux box, I had th
All you need is "/etc/init.d/iptables restart".
On 01/24/2010 04:01 PM, James Babiak wrote:
> I tried adding 5090 to the plugin, restarting firewall, and tested.
> Didn't work same 19.226.0.0 IP.
>
> Then I tried disabling the plugin altogether, restarting firewall, and
> tested. Still didn't wor
Ok, you're misunderstanding how the plugin works.
The signaling channel (SIP) terminates on your Asterisk box, and
Asterisk stays in the call for its duration.
5060 is the standard SIP port used by Asterisk (and most other SIP PBX's).
The plugin configures a netfilter connection-tracker to *also
On Jan 24, 2010, at 6:01 PM, James Babiak wrote:
> I tried adding 5090 to the plugin, restarting firewall, and tested. Didn't
> work same 19.226.0.0 IP.
>
> Then I tried disabling the plugin altogether, restarting firewall, and
> tested. Still didn't work with same result.
>
> I shouldn't nee
I tried adding 5090 to the plugin, restarting firewall, and tested.
Didn't work same 19.226.0.0 IP.
Then I tried disabling the plugin altogether, restarting firewall, and
tested. Still didn't work with same result.
I shouldn't need to restart the system for those changes to go into
effect, ri
Hey,
Yes, but only for UDP 5060, as this is the port that Asterisk is
listening on. I have 5090 configured for the ATA, but didn't enable it
in sip-voip.conf, figuring it's just being (supposedly) passed thru and
NAT'd.
Should I enable it for this port too or disable the plug-in altogether?
Have you enabled /etc/arno-iptables-firewall/plugins/sip-voip.conf ?
On 01/24/2010 01:11 PM, James Babiak wrote:
> Hey Everyone,
>
> I'm running into a weird issue, and hopefully someone can assist me in
> finding out what's going on.
>
> I'm running Astlinux 0.7 on a box serving as my router, a
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