asterisk users - May 2010 - Problems with Asterisk and two Linksys SPA941

Hi

I have a big problems on my Asterisk systems :

I have one Asterisk Server with static IP (no nat) and
6 Linksys SPA941.

All SPA are after a router with NAT:

     * SPA-1 and SPA-2 are on the same network,
        we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet
router

     * SPA-3,
        we have a pat 5062 => SPA-3

     * SPA-4,
        we have a pat 5063 => SPA-4

     * SPA-5,
        we have a pat 5064 => SPA-5

     * SPA-6,
        we have a pat 5065 => SPA-6

On the Asterisk Sip conf, we have nat=yes and dynamic host.


The problems are SPA-1 and SPA-2 can call to all other SPA except SPA-3
with SPA-3, i speak, it's good, spa-3 have the sound, but spa-3 speak
i don't have the sound.

SPA-3 can speak with SPA-4,5 and 6 without problems

a idea of the problems ?

Thanks
Bye

Do you have UDP 10000 to 20000 port forward to your router?

What kind of router is it?

Respectfully
Michael D Mosier
Ftoc Certified

On May 16, 2010 12:27 AM, "Olivier CALVANO" <o.calvano at
gmail.com> wrote:

Hi

I have a big problems on my Asterisk systems :

I have one Asterisk Server with static IP (no nat) and
6 Linksys SPA941.

All SPA are after a router with NAT:

    * SPA-1 and SPA-2 are on the same network,
       we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet router

    * SPA-3,
       we have a pat 5062 => SPA-3

    * SPA-4,
       we have a pat 5063 => SPA-4

    * SPA-5,
       we have a pat 5064 => SPA-5

    * SPA-6,
       we have a pat 5065 => SPA-6

On the Asterisk Sip conf, we have nat=yes and dynamic host.


The problems are SPA-1 and SPA-2 can call to all other SPA except SPA-3
with SPA-3, i speak, it's good, spa-3 have the sound, but spa-3 speak
i don't have the sound.

SPA-3 can speak with SPA-4,5 and 6 without problems

a idea of the problems ?

Thanks
Bye

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Hi,

I have the same setup as you.

I didn't bother mapping any ports.
I just enabled nat and keepalive.

Here is a screenshot of the config on the phones. For some reason, on some
phones I had to turn the NAT Mapping Enabled to Off otherwise call transfer
didn't work.

http://www.postimage.org/image.php?v=Tsz5WIJ

The sip.conf setup for this phone is:-

[winsor_202]
type=friend
context=winsor_phones
host=dynamic
secret=passwordhere
nat=yes
disallow=all
allow=gsm
allow=ulaw
canreinvite=yes
vmexten=1 at winsor
mailbox=1 at winsor
pickupgroup=1
callgroup=1

Hope that helps.
Dan