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)

Iīve  tried in my sip.conf with and without NAT=1.

In the phone, if I set the outbound proxy to the linksys it doesnīt 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).

Donīt know what else to try. Can someone please help me?


----- Original Message -----
From: "Greg Renouf" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
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: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Stephen R.
> Besch
> Sent: 05 June 2003 19:03
> To: [EMAIL PROTECTED]
> 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
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
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>
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