[asterisk-users] Help with remote users

Tue Oct 7 09:24:47 CDT 2008

I have just confirmed that they may be having a problem with double NAT.
They have two ATAs, and they have two different DSL connections.  One set-up
goes from the first DSL Modem (NAT & Wirless are disabled on the DSL Modems)
to a Linksys WRT110 and then there is a WRT54G hooked in to the 110 that has
the ATA plugged into it.

The other ATA is configured from a DSL Modem (again, I was told NAT &
Wireless were disabled on the modem) to a WRT600N and the ATA is plugged in
there. 

I have the same issues on both ATAs.  I have no idea why their network is as
poorly designed as it is, the bad part is I have to make sure the phones
work there and try to troubleshoot from 3000 miles away.

Any work arounds for a problem because of double NAT? A quick and dirty
solution for them to get their phones working right?

Steve Anness

On 10/7/08 2:12 AM, "Andrew Joakimsen" <joakimsen at gmail.com> wrote:

> Make sure they are not using double NAT. Many ISPs these days send
> their subscribers a "modem" that in reality is a router.
> 
> Also if you can post the PAP2 configuration. I hope you are using
> provisioning.. too bad Linksys makes it possible to obtain that
> information.
> 
> 
> On Mon, Oct 6, 2008 at 12:40 PM, Steve Anness <steve.anness at gmail.com> wrote:
>> I am using NAT so the ATAs are configured with a proxy server.  Qualify is
>> set to yes.  Here is what is happening.  After they plug in the ATA on the
>> otherside, and things register and I can call and they can call.  After
>> several minutes I try to call and then get the "no-service" message.  This
>> is with Qualify=yes.
>> 
>>    -- Executing [7193134525 at excel-in:1] Set("SIP/10.10.30.213-b7823fc0",
>> "CDR(accountcode)=Hiramine") in new stack
>>     -- Executing [7193134525 at excel-in:2] Set("SIP/10.10.30.213-b7823fc0",
>> "CALLERID(all)=(Hiramine) "" <2545239280>") in new stack
>>     -- Executing [7193134525 at excel-in:3] Dial("SIP/10.10.30.213-b7823fc0",
>> "SIP/17110-1&SIP/17112-1|20| w") in new stack
>> [Oct  6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to
>> create channel of type 'SIP' (cause 3 - No route to destination)
>> [Oct  6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to
>> create channel of type 'SIP' (cause 3 - No route to destination)
>>   == Everyone is busy/congested at this time (2:0/0/2)
>>     -- Executing [7193134525 at excel-in:4]
>> Playback("SIP/10.10.30.213-b7823fc0", "ss-noservice") in new stack
>> 
>> If qualify is equal to no, then it just trys to ring, I get no errors it
>> just keeps trying (except the phone doesn't actually ring).
>> 
>> I just wrote an email to find out more about their network settings there.
>>  To see if the ATAs are actually getting a private or public address.  If
>> they are getting a public address I suppose I can just set NAT=no and as
>> long as I can ping the public address and port 5060 isn't blocked by a
>> firewall than I should be able to resolve these issues.
>> 
>> Thanks for your time.
>> 
>> Steve Anness
>> 
>> 
>> 
>> On 10/6/08 2:20 PM, "Jerry Jones" <jjones at danrj.com> wrote:
>> 
>> 
>> On Oct 6, 2008, at 1:53 PM, Steve Anness wrote:
>> 
>> I know I have asked about this before, but I thought that I would ask again
>> with some more detail and maybe someone will have an idea.  This is my first
>> time to be setting up an asterisk server and I have a server running.  I
>> sent Linksys PAP2T's to several remote users.  Only one out of the four
>> users actually work like they should.  One of the other users I am assuming
>> is behind a firewall on his wireless router and needs to open up the proper
>> ports.  However, I have two users in New York on a DSL connection and I
>> can't understand why things are happening like they are.
>> 
>>  Here Is the situation.  Both users can plug in their ATAs and I can watch
>> the server output, they register and then they can make calls and I can call
>> them. Some time later (usually within minutes) the ATAs show to be
>> "unreachable" and I can no longer call; however, they can still make calls.
>> 
>> 
>> do you have qualify=yes ??
>> Is asterisk on a public IP?
>> 
>> 
>> 
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> 
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