[Asterisk-Users] Budgettone 100 phone Configuration

[William Zhang]

If your proxy server has the same DNS domain name as your SIP URI, then
leave outbound proxy blank.

Then, check "yes" for NAT Traversal. Type in the STUN server ip, GS has
a STUN server 67.153.142.67, or you can use vovida's 66.7.238.210.

BTW, there is a new release, check the web site:
www.grandstream.com/y-service.htm. You can configure your GS phone with
GS tftp server: 4.3.153.56, the reboot the phone. Since you use
1.0.3.53, you will lose you phone configuration this time, you will not
lose any confituration from 1.0.3.58 and later release.

--- Dan Fernandez <danfernandez00 at hotmail.com> wrote:
> Will  look into this once someone can help me with the configuration
> behind
> NAT (without NAT I have no problem)
> I am using v1.0.3.53 and a linksys router (the phone IP is
> 192.168.1.2)
> 
> I&#30939;e  tried in my sip.conf with and without NAT=1.
> 
> In the phone, if I set the outbound proxy to the linksys it
doesn&#30936;
> do
> anything. If I leave outbound proxy empty it registers and I can
> place calls
> but no audio either way. I have also tried setting the phone for NAT
> and no
> NAT (no STUN server).
> 
> Don&#30936; know what else to try. Can someone please help me?
> 
> 
> ----- Original Message -----
> From: "Greg Renouf" <grenouf at well.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Thursday, June 05, 2003 8:16 PM
> Subject: RE: [Asterisk-Users] Budgettone 100 phone Configuration
> 
> 
> > I'm using v.1.0.3.58 and am experiencing that my phone crashes
> every
> > time the call reaches about 45 minutes in length.
> >
> > Has anybody had a similar experience?
> >
> > -GSR
> >
> >
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Stephen
> R.
> > Besch
> > Sent: 05 June 2003 19:03
> > To: asterisk-users at lists.digium.com
> > Subject: re: [Asterisk-Users] Budgettone 100 phone Configuration
> >
> > The updated Budgetone firmware (1.0.3.60) has indeed fixed the
> "silent
> > DTMF" issue.
> >
> >  >By the way, Grandstream just got the "silent DTMF" problem fixed
> for
> > me
> >  >and sent me an updated revision this morning (1.0.3.60).  I am
> just
> >  >about to install it, but it may require that I debug my tftp
> server,
> >  >which I haven't tested yet.  I'll post the list as soon as I get
> the
> >  >new version loaded.
> > --
> > Stephen R. Besch, Ph.D.
> > SachsLab
> > 320 Cary Hall
> > SUNY at Buffalo
> > Buffalo, NY 14214
> > (716) 829-3289 x106
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > _______________________________________________
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> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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=====

William Zhang