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

[Brian Capouch]

Christopher J. Wolff wrote:
> Is it safe to assume that a fresh CVS build will not have the SIP
> translation problem described?
> 
> Regards,
> Christopher
> --__--__--

> 
> 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.

B.


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