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

[WipeOut]

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?
>>
>> 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.
>
>
Your ATA-186 and Bugetone are both behind NAT or both non-NAT??

In otherwords are they both in the same setup in relation to the server??

Later..