Have you tried using a recent trunk build, like 2104 with the Arno 
firewall enabled and the sip-voip plugin turned on?

-Philip


Ron Byer Jr. wrote:
> Greetings. 
>
> Thanks for the response!
>
> I am using 0.6.1. It's easy enough to try the latest 0.6.2 version, so I
> will pull that down.
>
> Single-homed. I do have a 2nd external interface, but I've taken it down to
> reduce the problem set at the moment. Static ip address. 
>
> Firewall configuration on the single EXTIP:
>
> EXTOPEN="t22 t8088 u4569 t80 u5060 u10000:20000"
>
>
> I've got to go back to the rtp traffic streams and make sure that the rtp
> port(s) are below 20000.
>
> Thanks again.
>  
> Ron Byer Jr.
> NetWeave Integrated Solutions, Inc.
> +1.732.786.8830 x120
>  
>
> -----Original Message-----
> From: Tod Fitch [mailto:[EMAIL PROTECTED] 
> Sent: Friday, November 21, 2008 12:35 PM
> To: AstLinux Users Mailing List
> Subject: Re: [Astlinux-users] nat
>
> I am experimenting with AstLinux 0.6.1 on a net5501 but the topology  
> is different in the my pbx is on the NATed LAN with most (but not all)  
> of the phones. I have not seen an issue with the net5501 failing to  
> handle the RTP stream, my issue is that on ENUM calls Asterisk is  
> attempting a "native bridge" with my NAT/firewall/router does not allow.
>
> Anyway, are you using 0.6, 0.6.1 or the latest 0.6.2 release candidate  
> which I think is labeled as astlinux-0.6-2082 on
> http://www.djhsolutions.com/astlinux/
>
> Also, are you running "single homed" or do you have multiple  
> interfaces online?
>
> How is the firewall on your AstLinux box configured?
>
> --Tod
>
> On Nov 21, 2008, at 8:58 AM, Ron Byer Jr. wrote:
>
>   
>> I've started using the astlinux 0.6 on a net5501 in earnest  
>> recently, and we have found ourselves in a world of hurt from a NAT  
>> perspective. We have a few phones behind their own soho firewalls,  
>> and the pbx is in the clear. 1-way audio, no-way audio  for these  
>> NATted phones seems to be the deal.  The SIP traffic works fine,  
>> even to the point of hangup. There just isn't any media stream.
>>
>>
>>
>> Until this is solved, I've set up another PC-based PBX, running  
>> asterisk on CentOS, just for the NATted phones It works fine. Same  
>> exact version of asterisk :: 1.4.21.2
>>
>>
>>
>> I've been debugging this for the past week, and have followed the  
>> bread crumbs into the poll loop in ast_waitfor_nands. I can flip  
>> between the two installs (astlinux 0.6 and CentOS) and track the  
>> debug messages (including a ton of my own at this point) all the way  
>> to expecting the first RTP packet. Both installs track exactly. The  
>> CentOS side then starts picking up the rtp packets and bridging the  
>> streams. The Astlinux side hangs in the poll until the hangup. The  
>> poll wakes up on the SIP BYE message, which is very interesting.   
>> Tcpdump shows the rtp packets banging into the box from the phone  
>> and the ITSP, but the poll never wakes up until the hangup.
>>
>>
>>
>> I came across some list mail from a few months box about one-way  
>> audio and NAT issues that would be solved(?) with an kernel update  
>> to .25 and a new uCLIBC. Not sure if that is wishful thinking or  
>> not. Also, my CentOS kernel version is 2.6.18, two points back from  
>> my astlinux 2.6.20.
>>
>>
>>
>> Any thoughts/insight would be appreciated. I'm fresh out of ideas at  
>> present.
>>
>>
>>
>>
>>
>> <image001.jpg>
>>
>>     


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