Hi George,
Do you have qualify=yes set in sip.conf for your phones?
When you check sip show peers, does it give you an OK (X ms) or does it
say UNREACHABLE or UNMONITORED?
If you enable qualify=yes or qualify=[some number] then Asterisk will
poll the SIP UA every once in a while to make sure it is still
reachable. This may or may not work. In some cases, if the UA doesn't
support the SIP OPTIONS correctly, it will come back and Asterisk will
think it is unreachable until it sends another register command. In
other cases, it helps keep the ports open on the firewall.
BTW, we have successfully tested NAT with multiple user agents as you
describe with pretty much plug and play with Linksys, SMC,
Shorewall/Linux and various other NAT router/fw devices with great
success. Thus far, we've only had problems with DrayTek routers
mangling the UDP packets. In those cases, the UAs registered
successfully and all inbound calls worked, but outbound calls did not
as the UDP/RTP streams weren't getting handled correctly by the router.
They have an updated firmware that solves this problem, but we haven't
finished testing it.
On Wednesday, August 13, 2003, at 09:25 PM, Adams, Gavin wrote:
>> From: George Lin [mailto:glin@cosini.com]
>>
>> I want to deploy multiple SIPs phone in our office. And we have
> shutdown
>> the
>> firewall at our office router(with ip 211.x.x.x). we have deployed the
>> asterisk with IP 218.x.x.x.
>>
>> All SIP phones have 192.x.x.x.
>
> We have something similar George, * sits outside the firewall with a
> registered IP address, the SIP phones sit behind the firewall with
> 172.16.x.x addresses.
>
>> When the SIP phone is power on, they are registered in the asterisk.
> we
>> can
>> check at asterisk side by issueing "sip show peers", and all
the
> phones
>> are
>> associated with 211.x.x.x:port-number.
>
> Sounds familiar. Question, do you hide all the phones behind a single
> IP
> address, or does each phone get a unique address? Also, what type of
> firewall?
>
>> PRoblem:
>> Now some times the sip can receive call, and some time it cannot
> recieve
>> call. When we dumping the sip log, and see that asterisk tried to
> INVITE
>> the
>> specified SIP phone with the 211.x.x.x:port-number, and was failed
> after 5
>> times. But the call orginated from SIP phone is always OK.
>
> Yup, what we initially found. Basically, we started by attempting to
> hide all the phones behind a single IP address. In this case, make sure
> you uniquely assign the control port (by default UDP 5060) to something
> different for each phone.
>
> We use FireWall-1 (older version) that doesn't play nice with
"hide
> NAT". Basically, it would timeout UDP connections after 40 seconds of
> no
> activity. Not good unless you reduce the reregister time to something
> crazy like 30 seconds. Check to see how your firewall/NAT device
> handles
> [P]NAT translation.
>
>>
>> Questions are:
>>
>> 1. Does asterisk remember the mapping between 192.x.x.x AND
>> 211.x.x.x:port-number ?
>
> It shouldn't. It might see the 192.x.x.x address in the SIP
> conversations, but even if it did, it would not be able to route the
> packets back.
>
>> 2. When a call to a sip phone, is it asterisk responsiblility to map
> the
>> 211.x.x.x:port-number to the 192.x.x.x, and send to the office router
> ? OR
>> it is the office router to remeber all the mapping between each sip
> phone
>> 192.x.x.x and 211.x.x.x:port-number, and asterisk juts sends the
>> 211.x.x.x:port-number to the office router ??
>
> Asterisk should attempt to contact the phone based upon the IP and port
> seen during a 'sip show peers'. Network device responsible for any
and
> all translations.
>
>> 3. If it is the office router's responsiblity, what should we
> configure
>> the
>> office router even there is no firewall???
>
> Unsure about this, I'd focus more on the NAT device. Can you describe
> the topology from the SIP phone to *?
>
> Regards,
>
> --- Gavin
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