asterisk users - Mar 2003 - SIP and NAT - more

Oh, and yes, the * is current as of a few days ago .. so it should have 
that new SIP code mark was working on a while back.

Thanks

have you tried nat=1 in your friend declaration?  I notice in your dump it
says "non-NAT"

Mark

On Fri, 21 Mar 2003, denon wrote:
> Oh, and yes, the * is current as of a few days ago .. so it should have
> that new SIP code mark was working on a while back.
>
> Thanks
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

Thanks -- I didn't realize that needed to be set.  It works now, but 
there's a horrible echo on the sip client side. (I dont know about the 
other side, as I havent called any humans yet :)

I don't, however, hear an echo when I call voicemail or such .. so I'm 
assuming it's something with the bridging?

I didn't know of any echo cans that need to be enabled for sip - are there? 
The PSTN line its connecting out on has echocan and whenbridged enabled.

Here's an example of one of the pstns, they're all built the same, using
an
Adtran 750 channel bank with current firmware (actually, the last release, 
which was considered the most stable by most):
context => pstn1
signalling => fxs_ks
amaflags => documentation
echocancel=yes
echocancelwhenbridged=yes
adsi=yes
channel => 17


Ideas? Thanks

At 09:53 PM 3/21/2003 -0600, you wrote:
>have you tried nat=1 in your friend declaration?  I notice in your dump it
>says "non-NAT"
>
>Mark
>
>On Fri, 21 Mar 2003, denon wrote:
>
> > Oh, and yes, the * is current as of a few days ago .. so it should
have
> > that new SIP code mark was working on a while back.
> >
> > Thanks
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users

On Fri, 21 Mar 2003, Mark Spencer wrote:
> have you tried nat=1 in your friend declaration?  I notice in your dump it
> says "non-NAT"
>
I?m in the same situation, trying to debug an ATA 186 behing a NAT.
And i?m stuck with "SIP/2.0 407 Proxy Authentication Required" debug
messages. Does anyone have any hints on thisone?

It would also be nice if someone could post a working ATA186 config. This
would help against stupid mistakes in that end.


But back to the NAT/No-NAT issue. What exactly is the difference
protocolwise inbetween the two of them?

	/Chris