[Asterisk-Users] RE: SIP i.e. Is something broken?

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This is my issue as well, Does anyone know how to fix it?

Thanks,
Michael

On Mon, 29 Sep 
2003, Dave Weis wrote:

> 
> On Mon, 29 Sep 2003, Brian Capouch wrote:
> > Christopher J. Wolff wrote:
> > > Is it safe to assume that a fresh CVS build will not have the SIP
> > > translation problem described?
> > > 
> > > Just FYI: I had similar problems for a while, and then I completely 
> > > scrapped my CVS directory and did a CVS CHECKOUT (instead of an update). 
> > > That solved the problem.
> > 
> > I "checkout" rebuilt from CVS last night about 10:00 pm, and filed a bug 
> > report at that time.
> > It isn't *all* SIP calls for me, btw; my ATA186 works just fine, but my 
> > Budgetone won't work with the "broken" code. . .
> > It's as described in other mails--I can receive calls on the Budgetone 
> > but when I make them the RTP part is broken and the calls cut off the 
> > second they're set up.
> 
> I'm seeing the same thing with my budgetones and today's cvs. They are all 
> on the same network and worked on previous versions. I can call 
> voicemailmain and the console says that it is playing but I hear no sound. 
> Then it hangs up automatically.
> 
> 
>