[asterisk-users] Do I need a SIP Proxy for this?

[Jonathan Moore]

On Wed, May 20, 2009 at 1:50 PM, Tim Nelson <tnelson at rockbochs.com> wrote:
> Could you elaborate a bit more?
> What isn't 'working out to well'?
> Are you getting failed calls? One way or no audio?

Sorry for the lack of information. I posted in a bit of haste.

Initially it was failed calls, or not being able to register.  I had a
line similar to register => 00000 at proxy01.sipphone.com in sip.conf and
it was never able to successfully register.  I would get a timeout
after so long, and then it would send again.

I then added the externalip and localnetwork configurations to
sip.conf and set the proxy01.sipphone.com section to include the
nat=yes, and this netted me one way audio, only after i swapped out
the aging cisco router for a vyatta install.

I mostly followed guides found on voip-info.org for gizmo and sip, and
also the information on Gizmo's website.

Another area that had issues with with having something like
Dial(SIP/remotehost) would fail to connect to remotehost.

-jonathan