[Asterisk-Users] Sipura SPA-2000 & Asterisk 1.24 w/incoming calls

voiplist gotvoip at gmail.com
Sat Jun 17 15:52:03 MST 2006


Yes, I have limited access to one SPA-2000 at the moment.

Anyone else seeing this?

When you say open a bug on this do you mean with Asterisk or Sipura?

I guess that's part of the problem, not sure if we should be
troubleshooting on Asterisk or the Sipura device..



On 6/17/06, Rich Adamson <radamson at routers.com> wrote:
> voiplist wrote:
> > We have issues with all of the SPA-2000 ATAs we have where incoming
> > calls from only one of our Asterisk servers do not complete.
> >
> > Details:
> >
> > 1- On the CLI we see that when the call is pushed to the ATA it shows
> > Busy/Congested
> > 2- We can make calls to this same server just fine
> > 3- We can receive calls from other Asterisk servers running older CVS
> > versions of Asterisk with the same exact ATA configuration on the same
> > exact network, behind the same exact NAT routers.
> > 4- We have tried putting the ATAs in front of the NAT routers using
> > the DMZ setting, no help.
> > 5-The ATAs do register just fine
> > 6- This happens on multiple networks behind multiple different NAT routers
> > 7- We have tried turning off the firewall on the server side temporarily
> > 8-We have tried a Grandstream 486 at one of these locations and all is
> > well with receiving incoming from this server
> >
> >
> > Remember, these ATAs work fine currently at their current locations on
> > their current networks, behind their current NAT routers when
> > communicating with a different Asterisk server.
> >
> > Other ATAs in these same locations have no trouble receiving calls
> > from the newer server.
>
> Interesting... I just ran into the same problem with no firewalls or
> nating involved. This one is an spa3k but with exactly the same issue.
> Had been working just fine, but now fails with SVN-branch-1.2-r34400.
> All other sip phones are functioning fine.
>
> An ethereal trace only indicates the spa3k is returning "busy here". I
> don't see anything wrong with the initial sip packets, but didn't have
> the time to dig to deeply into it either.
>
> I also made sure DND, Call Forwarding, etc, was not the issue.
>
> Looks like we need to open a bug on this. I'll be out of town for the
> next week and won't have any access to the boxes for testing.
>
> R.
>
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