[Asterisk-Users] SIP and NAT - more

[denon]

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
> >
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> >
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